2010:150 CIV MASTER'S THESIS Engine Noise Simulatorltu.diva-portal.org/smash/get/diva2:1031701/FULLTEXT01.pdf · 2016. 10. 4. · 2010:150 CIV MASTER'S THESIS Engine Noise Simulator

2010:150 CIV

M A S T E R ' S T H E S I S

Engine Noise SimulatorCreating Software for Controlling the Simulator

Including a Graphical User Interface, Calibration of the ENSand Transfer Function Measurements

André Lundkvist

Luleå University of Technology

MSc Programmes in Engineering Arena, Media, Music and Technology Department of Human Work Sciences

Division of Sound & Vibrations

2010:150 CIV - ISSN: 1402-1617 - ISRN: LTU-EX--10/150--SE

Engine Noise SimulatorCreating Software for Controlling the Simulator Including A Graphical User Interface,

Calibration of the ENS and Transfer Function Measurements

Andre Lundkvist

[email protected]

Sodertalje, Scania, October 1, 2010

II

Abstract

The purpose of the thesis was to create a graphical user interface (GUI ) which can controlspecific hardware for an Engine Noise Simulator (ENS ). The ENS should also be calibrated,and the software should be able to utilise these measurements to control the output soundpower level.

An ENS is an useful tool to investigate sound transfer paths and to simulate differentengines quickly without the need to replace the engine in the cabin.

The Scania ENS built at Lulea University of Technology, Lulea, uses 29 loudspeakerswith a surface microphone in each individual loudspeaker cavity for calibration purposes.There are loudspeakers on each side of the engine.

Software with a Graphical User Interface (GUI ) has been written usingMatlab’s GUIDE,to create a functional application that is easy to set up and use.

The GUI has many options so the user can specify how every loudspeaker should perform.The user can create sound groups, and assign any number of loudspeaker to the group,using individual phase and level settings. The types of sounds can be sinusoidal, noise(both white (Gaussian) and pink), sine sweep (or chirp) and also imported .WAV-files.

The GUI also handles all calibration functions, and has functions to verify the conditionof the ENS.

Programming of the software was done using Matlab, and includes functions to controleach loudspeaker individually, create as many different sounds as it’s needed and acquirefeedback data from each loudspeaker.

The ENS was calibrated by measuring transfer functions between sound power level andvolume velocity (source strength) of each loudspeaker element. The volume velocity iscalculated from measured pressure inside the loudspeaker cavities. By utilising feedbackfrom the surface microphones located inside each loudspeaker cavity, the user can specifya required sound power level for the output. The calibration was made with the LMSsuite.

III

IV

Preface

This Master of Science thesis has been carried out at Scania, located inSodertalje. The project was performed under supervision of the group RTTA,Truck Chassis Development and the department of Sound and Vibration atLulea University of Technology.

I would like to take the opportunity to thank Tony Algarp (RTRN), RogerJohnsson (LTU) and Ragnar Glav (RTRN) for guidance, support and the op-portunity to do this masters thesis at Scania. Thanks to Per-Olof Berglund(RTRN) for interesting conversations, tips and functions for Matlab. Bigthanks to all the people at Scania, for always being friendly and making metruly feel as a part of the team.

Very deep gratitude to my family and girlfriend, who has always supported mewith love (and money) during the hard times.

V

VI

Contents

1 Introduction 1

1.1 Background . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1

1.2 The Engine Noise Simulator . . . . . . . . . . . . . . . . . . . . . . . . . . . 2

1.3 Scope . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3

2 ENS Calibration 5

2.1 Theory . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5

2.2 Calibration Measurements . . . . . . . . . . . . . . . . . . . . . . . . . . . . 10

2.3 Calibration Function . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 12

2.4 Results . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 12

3 Software Development 17

3.1 Theory . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17

3.2 Database for Project Settings . . . . . . . . . . . . . . . . . . . . . . . . . . 18

3.3 Main GUI Window . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 19

3.4 Check/Import Calibration . . . . . . . . . . . . . . . . . . . . . . . . . . . . 21

3.5 Group Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 22

3.6 Trigger Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 25

3.7 Assign Groups . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 26

3.8 Adjust Levels . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 27

3.9 Acquire Input Signals . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 29

3.10 Data Preparation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 29

3.11 Comments . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33

4 Discussion 35

5 Future Work 37

VII

VIII CONTENTS

References 39

Appendix 1

A Loudspeaker Cavity Pressure 1

B Loudspeaker Volume Velocity 1

C Calculated Calibration Functions 1

D Measured Sound Power Levels 1

E Loudspeaker Directivity Test 1

F Measurement Instrumentation 1

G Manual 1

G.1 Quick Start . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2

G.2 Main GUI Window . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 9

G.3 Import Calibration Window . . . . . . . . . . . . . . . . . . . . . . . . 16

G.4 Check Calibration Window . . . . . . . . . . . . . . . . . . . . . . . . . 17

G.5 Edit Sound Groups Window . . . . . . . . . . . . . . . . . . . . . . . . 21

G.6 Loudspeaker Assignments Window . . . . . . . . . . . . . . . . . . . . 25

G.7 Trigger Settings Window . . . . . . . . . . . . . . . . . . . . . . . . . . 28

G.8 Adjust Output Levels Window . . . . . . . . . . . . . . . . . . . . . . . 31

G.9 Data Preparation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 37

G.10Running the ENS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 40

G.11Default Settings of the ENS . . . . . . . . . . . . . . . . . . . . . . . . . 43

G.12Calibration Files Specification . . . . . . . . . . . . . . . . . . . . . . . 46

Chapter 1

Introduction

1.1 Background

To meet the demands of customers and society, acoustic shielding is used to dampen thesound from a vehicle’s noise sources, such as engine and gearbox. A good noise encapsu-lation needs good material properties. An effective complement to numerical methods isto simulate the sound properties of a combustion engine by using a synthetic engine, witha large number of loudspeakers.

An Engine Noise Simulator (ENS ), is a valuable tool for investigating airborne noise trans-mission between the engine and passenger compartments [1] and the exterior. An ENS,compared to a single sound source, is capable of representing a realistic power unit. It cantherefore be used for evaluation and comparison between different bodies, or cabins, andmeasuring the effect of acoustical treatments and materials used in the construction.

There are many different types of sounds radiating from an engine. These can be splitinto several groups, such as combustion noise, idle noise and vibration, engine start-upand shut-down shake, air intake and exhaust noise, transmission noise and exterior noise(stationary and pass-by). There are also noise from auxiliary cabin heaters. [2]

When driving at low speeds, the engine combustion noise is the main part of noise inflictedon the driver and passenger, as the wind and road noise are minimal. Because of existingand future emission regulations for the engine, the margins for calibrating the combustionis getting smaller. This gives that a key of lowering engine combustion noise is within theattenuation provided by the cabin and chassis, which is mainly provided by the acousticalabsorption properties and sealing of the body.

Low frequency vibrations and booming sounds is a problem that can be generated by anidle engine. This is mainly a powertrain problem due to its mounting system. The pow-ertrain mounting needs very high isolation to the vehicle body to reduce these vibrations.Vibration and structure borne noise can be a significant problem in diesel engine vehicles,mainly due to the large dynamic force levels on the driveline and powertrain mountingsystems from the high low speed torque characteristics. [2]

In addition to the combustion noise, most diesel engines are equipped with turbochargerswhich give an increasing risk of high pitch noise. The increased gas flow through theexhaust system also gives turbulence noise. The turbulence generated can also spreadthrough the vehicle as structure borne noise.

1

2 CHAPTER 1. INTRODUCTION

In a comparison between a diesel engine and a gasoline engine, diesel engines have usuallyhigher external noise levels. A vehicle’s pass-by noise performance is highly affected bythe intake and exhaust systems. Gas rush, due to the turbocharger, is a common exteriornoise problem, and the lack of a throttle on the intake system, it can occasionally deliverhigh noise. [2]

1.2 The Engine Noise Simulator

The Scania Engine Noise Simulator (ENS ), designed and constructed at Lulea Universityof Technology, Lulea, is a model of an internal combustion engine, consistent of 29 loud-speaker elements located on all sides of the ENS. There are 6 elements on the top, 2 at thebottom, 2 at the front and 3 at the rear end. The left and right sides contain 8 loudspeakerelements respectively. See figure 1.1 for an overview of the engine. Inside each loudspeakercavity is a surface pressure microphone, which can be used for calibration purposes. TheScania ENS is very advanced and each channel can be controlled individually.

Figure 1.1: The design and construction of the Engine Noise Simulator

1.2.1 The Hardware Rack

The ENS loudspeakers are controlled from an external hardware rack (see figure 1.2 (b)),containing five 700 W Rotel power amplifiers and a computer with the required Digital Sig-nal Processing (DSP) hardware. The DSP hardware consists of two interconnected DataAcquisition Processors (DAP 5016a) [3], with isolated analog outputs using six MSXB076 cards. Analog inputs are provided with an analog back plane system and an externalConstant Current Power (CCP, or ICP) amplifier. The analog back plane is powered bythe Data Acquisition Processors. To control the Data Acquisition Processors, Matlab [4]is used.

1.3. SCOPE 3

The feedback signals are connected to the backplane cards by an external ICP amplifier.There are 32 inputs, but only 29 are used. The connections on the back side of the powerrack can be seen in figure 1.2.

(a) (b)

Figure 1.2: Connection of the feedback signals (a) and the ENS loudspeakers (b).

1.3 Scope

One part of the thesis is to create a Graphical User Interface (GUI ) to give the operatoran easy way of controlling the ENS.

The objective for the GUI is to have an easy way of controlling the Data AcquisitionProcessors, by using project files which can be saved or loaded, import calibration mea-surements and freely set up sounds for output. The main window should have the abilityto run the ENS in a calibrated setup to check that everything is working correctly.

There should be an option to send out as many different sounds as there are loudspeakergroups. A loudspeaker group can be defined as a list of loudspeakers, using the same soundsource. The grouping of the loudspeakers should be free, meaning that a loudspeaker groupcan contain as many loudspeakers as there are available. Any loudspeaker should be ableto be used as a reference, and there should also be the ability to set a phase for anyloudspeaker, either by a numeric value, slider, or ± sign change.

The sounds for a loudspeaker groups should be easy to specify, and there should be theoption to use basic signals such as sinusoid and noise, but also to import a .WAV file. Thereshould be adjustable parameters for each sound type, both by sliders and numeric values.

Levels from the feedback signals, gathered from the internal microphones inside eachloudspeaker cavity, should be easy to read in the interface. A picture of the ENS mightbe helpful for error determination, indicated by no feedback signal, or divergence from anominal level for the calibrated case.

The ENS should also be able to handle trigger signals, both send and receive. The outputtrigger from the ENS should be able to be sent upon start and/or stop, with pre and/orpost function. The input trigger should be able to start and/or stop the ENS operation.

4 CHAPTER 1. INTRODUCTION

When the ENS has a working GUI, it will be calibrated. The calibration function willgive the operator the option of setting a reasonable output sound power level for eachloudspeaker element.

With the GUI and the calibration functions, the ENS is in a working condition and shouldbe evaluated with either pass-by noise measurements or by modelling a combustion engineusing recorded sounds from an engine test cell.

In addition to the scope, there was also a manual written for the GUI. The manual canbe found in appendix G – G.12.

1.3.1 Scope Limitations

The Graphical User Interface is limited in a way that only one project can be opened ata time, and only the basic tools are available for the user. More sound processing optionscould have been implemented, but to limit the project in time and complexity, this waschosen not to be included in the GUI. This should not be a problem, as the user canprocess the imported files beforehand.

For the calibration functions, all loudspeakers are considered as monopoles, thus assumingthat the calculated volume velocity of the loudspeakers is valid for all frequencies. Thisis however only the case for very low frequencies. But since the transfer functions aremeasured as sound power to the volume velocity, this should not matter.

For the calibration measurements, the measurement sphere is not large enough to fulfilthe ISO standard 3745 [5] due to equipment availability. It is however enough for afirst calibration procedure, as future updates can be done to enhance the calibrationmeasurements.

Due to time limitations, the final measurements and modelling were not performed.

Chapter 2

ENS Calibration

2.1 Theory

2.1.1 Loudspeaker Elements as Monopoles

A monopole is the simplest form of a sound source and is considered as a point. Fromthe point, it produces a sound field that expands spherically [6]. The sound field can beexpressed at any distance from the origin by the volume velocity Q0 and the wave number,k, by the equation,

pm =ρ0iωQ0

4πre−ikr (2.1)

The wave number is simply defined as k = ωc , or k = 2πf

c , where ω is the angular frequency(rad/s), c is the speed of sound (m/s) and f is the frequency specified in Hz (1/s).

The sound power created by the monopole is given by,

Wm =ρ0ck

2

4πQ2

0 (2.2)

where ρ0 is the density of air. This gives that the sound power with respect to the volumevelocity (Q0) squared is a linear system.

For a loudspeaker to be able to be classified as a monopole, the wavelength of the radiatedsound must be much longer than the loudspeaker dimensions, including the mounting box.This is indicated by a so called Helmholtz number, or He-number, which is defined as thewave number multiplied by the largest dimension of the surface,

He = kl =2πl

λ(2.3)

The Helmholtz number should be small for monopole classification. For a loudspeakerwith a radius of 6.5” (16.5 cm), and a given Helmholtz number of 1, the highest frequencyshould be much lower than,

fmax ≪c

2π · 0.165≈ 330 Hz (2.4)

5

6 CHAPTER 2. ENS CALIBRATION

When the speaker can no longer be classified as a monopole, the directivity index DI, isgiven by,

DI = 10 log

[

4π∫ 2π0

(∫ π0F 2(θ, φ) sin θ dθ

)

dφ

]

(2.5)

where F (θ, φ) is a directivity function, describing the loudspeaker direction properties inthe distant sound field [7].

The sound intensity in the distant field from a loudspeaker can be calculated with,

I =ρ0ω

2v2ka4

8r2cF 2(θ, φ) (2.6)

where vk is the vibration speed of the loudspeaker membrane, a is the loudspeaker elementradius and r is the distance from loudspeaker.

2.1.2 Calibration Theory

Because the human hearing and perception of sound isn’t linear, it’s very common to refersound power and sound pressure in decibels (then called sound power level and soundpressure level). This is a logarithmic scale defined to give a more realistic description ofthe dynamic range [6].

For a given sound pressure, in pascals (Pa), the corresponding sound pressure level, Lp, iscalculated by,

Lp = 10 log

(

p

p0

)2

dB or Lp = 20 log

(

p

p0

)

dB (2.7)

A similar conversion is performed for sound power to sound power level, by using theequation,

Lw = 10 log

(

W

W0

)

dB (2.8)

For these conversions, the 0 dB reference levels for sound pressure and sound power are,

p0 = 2 · 10−5 Pascals (2.9)

W0 = 10−12 Watts (2.10)

which is the threshold for human hearing at 2500 Hz.

The sound pressure inside a loudspeaker cavity can be converted to volume velocity. Thisconversion can be described by a transfer function, here called Hq,k, which would give thevolume velocity. The index k indicates which loudspeaker of the ENS that belongs to thisindividual transfer function. As described in section 2.1.1 on page 5, the volume velocitysquared is in proportion to the radiated sound power of a monopole. The conversion fromsquared volume velocity Q2

k to sound power is therefore given by transfer function Hc,k.

Given this, and the transfer function between output signal and the produced pressure(Ha,k), the sound transfer chain from the output voltage signal to radiated sound powercan be modelled as,

vout,ky

Ha,kpv,k

y

Hq,kQk → Q2

ky

Hc,kWk or Wk = Hc,k (Ha,kHq,kvout,k)

2

(2.11)

2.1. THEORY 7

If assuming adiabatic behaviour in the loudspeaker cavity and also assuming that thefrequency of the acoustic pressure fluctuation has a wavelength that is much longer thanthe cavity dimensions (section 2.1.1), the volume velocity can be calculated from themeasured pressure by using the adiabatic gas law [6],

(p0 + pv,k)Vγk = constant (2.12)

where p0,k is the overall air pressure, pv,k is the RMS value of the acoustic pressuregenerated by the loudspeaker membrane and Vk = V0,k+Vk(t) is the volume of the cavity.

By differentiating the gas law, with respect to the time, t, the change of volume inside thecavity over time can be extracted. Equation 2.12 in differentiated form is given by,

V γk

δpv,kδt

+ (p0,k + pv,k)γVγ−1k

δVk

δt= 0 (2.13)

Assuming there are only small fluctuations, this expression can be simplified. By usingthe definition of γ = (c2ρ0)/V0,k removing the stationary entities and using the Fouriertransform, this can be simplified as,

pv,k = −c2ρ0V0,k

Qk (2.14)

The source strength can therefore be calculated by multiplying the measured air pressurelevel, Lpv,k (or pressure pk) with a constant that depends on the speed of sound in air (c),volume of the specific loudspeaker cavity (V0,k) and the density of air inside the cabinet(ρ0).

Qk =V0,k

c2ρ0iω

√

10Lpv,k/10pref or Qk =V0,k

c2ρ0iωpk (2.15)

The transfer function from measured pressure to volume velocity is therefore,

Hq,k =V0,k

c2ρ0iω (2.16)

Since the sound power level is linear to the volume velocity squared, a useful calibrationfunction for loudspeaker element k could thus be given by,

Hc,k =Wk

Q2k

or Wk = Hc,kQ2k (2.17)

As previously described, the volume velocity Qk can be measured by using the surfacemicrophone inside the loudspeaker cavity. By using the transfer function Hc,k and themeasured volume velocity, the expected sound power level can be calculated. In otherwords, the output amplitude to the power amplifier could be adjusted to give a certainvolume velocity which produces a wanted sound power level.

Because the user should specify a requested sound power level output, the pressure neededinside the cavity can be calculated by replacing the volume velocity Qk with it’s definition.

pk =

√

Wk

Hc,k

(

c2ρ0iωVk

)

(2.18)


2.1.3 Measurement Theory

According to ISO standard 3745:2003 there should be an array of 20 microphones formeasurements in an hemi-anechoic room [5]. This number of microphones is sufficient ifthe difference in dB between the highest and lowest SPL measured at any position and inany frequency band of interest is numerically less than the half the number of microphonepositions.

The ISO standard 3745:2003 also specifies other measurement methods, not using a mountedhemisphere. These are by using a single microphone and sweep the surface over multiplecircular paths, spiral path or meridional arcs. Since Scania has a measurement hemi-sphere, this method was used.

The maximum allowed difference between repeated measurements can be found in ta-ble 2.1. This reproducibility has to be fulfilled for valid measurements [5].

Table 2.1: Estimated upper values of the standard deviation of sound power levels’ andsound energy levels’ reproducibility.

One-third-octave mid band frequency Upper values of std of reproducibility, σR

(Hz) (dB)Anechoic room Hemi-anechoic room

50 to 80 2.0 2.0100 to 630 1.0 1.5800 to 5000 0.5 1.06300 to 10000 1.0 1.512500 to 20000b 2.0 2.0

A-weighted 0.5 0.5b If the instrumentation allows and if correction is made for absorption of sound by the atmosphere.

The sound power level can be calculated from the 20 microphones, using the surface soundpressure level Lpf , with the equation [5],

Lw = Lpf + 10 log(

2πr2)

dB + C1 + C2 (2.19)

where r is the radius of the hemisphere. The energy-average of the time-averaged soundpressure levels at all the microphone positions on the measurement surface (Lpf) is definedas,

Lpf = 10 log

(

1

N

N∑

i=1

100.1Lpi

)

dB (2.20)

where Lpi is the sound pressure level corrected for background noise from the i:th micro-phone position in dB, and N is the number of microphone positions.

To compensate for weather conditions, the two entities C1 and C2 is expressed as,

C1 = −10 log

(

B

B0

√

313.15

273.15 + θ

)

dB (2.21)

C2 = −15 log

[

B

B0

(

296.15

273.15 + θ

)]

dB (2.22)

where B is the barometric pressure during the measurements (in Pascal). B0 is the refer-ence barometric pressure (1.01325 · 105) and θ is the temperature. Equation 2.22 is only

2.1. THEORY 9

valid for the temperature range 15 oC ≤ θ ≤ 30 oC. Humidity differences can be ignored,as the maximum correction for this is ≈ 0.04 dB [5].

Figure 2.1: Microphone positions on the hemisphere seen from the top (left figure) andside (right figure). [5]

Table 2.2: Microphone positions for hemisphere with r = 2.7 m, in hemi-anechoic room

Microphone Normalised Position Absolute Position (m)Number x/r

y/rz/r x y z

1 -1.00 0.00 0.025 -2.70 0.00 0.0675

2 0.50 -0.86 0.075 1.35 -2.32 0.2025

3 0.50 0.86 0.125 1.35 2.32 0.3375

4 -0.49 0.85 0.175 -1.32 2.32 0.4725

5 -0.49 -0.84 0.225 -1.32 -2.27 0.6075

6 0.96 0.00 0.275 2.59 0.00 0.7425

7 0.47 0.82 0.325 1.27 2.21 0.8775

8 -0.93 0.00 0.375 -2.51 0.00 1.0125

9 0.45 -0.78 0.425 1.22 -2.11 1.1475

10 0.88 0.00 0.475 2.38 0.00 1.2825

11 -0.43 0.74 0.525 -1.16 2.00 1.4175

12 -0.41 -0.71 0.575 -1.11 -1.92 1.5525

13 0.39 -0.68 0.625 1.05 -1.84 1.6875

14 0.37 0.64 0.675 1.00 1.73 1.8225

15 -0.69 0.00 0.725 -1.86 0.00 1.9575

16 -0.32 -0.55 0.775 -0.86 -1.49 2.0925

17 0.57 0.00 0.825 1.54 0.00 2.2275

18 -0.24 0.42 0.875 -0.65 1.13 2.3625

19 -0.38 0.00 0.925 -1.03 0.00 2.4975

20 0.11 -0.19 0.975 0.30 -0.51 2.6325

When measuring with an hemisphere in an hemi-anechoic room, there are some require-ments for the radius of this hemisphere. The radius should be equal to, or greater than, 1m. However, if the lowest frequency of interest has a λ/4 that is larger than 1 m, this isthe minimum radius. The radius must be larger than twice as long as the largest sourcedimension, or three times the distance of the source’s acoustic centre from the reflectingplane [5]. The radius of the hemisphere used for the following measurements is 2.7 m.

The microphone positions can be a predetermined setup given in the standard, or a user-defined arrangement that meets the requirements. The microphone positions used in the


calibration procedure can be found in figure 2.1. Table 2.2 gives the defined microphonepositions in relation to the radius r, but also the absolute positions for the case thehemisphere has a radius r = 2.7 m. The number of microphones used is enough if thehighest measured sound pressure level difference is below half the number of microphonesin dB. If this is not fulfilled, the source can be rotated 180 degrees around the z-axisduring measurement.

All microphones shall be calibrated with a sound calibrator of class 1 accuracy. It is alsoimportant that all air ducts, electrical conduits and piping connected to the source don’tradiate a significant amount of sound energy. If possible, all auxiliary equipment used bythe source that is not part of the source should be placed in an outside location of the testroom.

2.2 Calibration Measurements

The measurements were performed with guidelines from the ISO standard 3745, revisedyear 2003 [5]. The sound power level from each loudspeaker element was measured usinga hemisphere in an hemi-anechoic room at Scania (the room Chassie Dynamometer 4 ).The engine was mounted with rubber bushings on a metal frame. Each element of the ENSis used as a source one at a time.

For sound power level measurements, the ENS was located in the center of the sphere.This is displayed in figure 2.2. The sphere containing the ENS was located in the frontcenter part of the room. The center loudspeakers on the left and right side are alignedwith the microphones on the x axis on the sphere. The sphere y axis is aligned with thefront and center back speakers. This will make the rear of the engine beam sound towards-y and the left towards +x directions.

(a) (b)

Figure 2.2: Overview of the calibration measurement setup with views from the front ofthe engine (a) and from the side of the engine (b)

Because the measurement frequency range extends below 160 Hz, the measuring intervalneed to be at least 30 s. The measuring time used for each measurement is set to 1 s,and the mean sound pressure level is calculated from 120 measurements, giving a total of120 seconds. The background level need to be at least 10 dB below the measured soundpressure levels.

The settings used for the measurements can be found in table 2.3. The measurement time

2.2. CALIBRATION MEASUREMENTS 11

is long enough for a 1 Hz resolution when using a bandwidth up to 8192 Hz. White noisewas chosen to represent a full spectrum for the calibration functions.

Table 2.3: Calibration Measurement SettingsI/O Setting Value

Overall Bandwidth 8192 HzResolution 1 HzSpectral Lines 8192Measurement Time 120 s (total)Number of Means 120Time Window 1 s (mean)Means LinearWindow Hanning

Output Type White NoiseAmplitude 1 V

Input Type AutoPowerSpectrumAmplitude Scaling RMSSpectrum Unit PaSpectrum Format LinearWeighting None

No correction for background noise was made, because the difference between backgroundnoise and measured sound pressure levels was much higher than 20 dB for the lowerfrequency range, as stated in section 2.1.3.

The instrumentation used for the measurements can be found in appendix F, All mea-surements were made in an environment with a reflective floor, which has to be taken intocondsideration when analysing the directivity drawings, which can be found in appendix E.

The acoustic environment that was present during the measurements can be found intable 2.4.

Table 2.4: Acoustic Environment in CD4

Specification Data

Temperature 23 degrees Celsius

Humidity 3 %

Atmospheric Pressure 1032 Pa


2.3 Calibration Function

The calibration function is defined according to section 2.1.2, using equation 2.23. Thereare one unique calibration function for each loudspeaker on the ENS. The sound power levelswas calculated using equation 2.19, then converted to sound power by inverse logarithmiclaw.

Hc,k =Wk

Q2k

(2.23)

The volume velocity was calculated using equation 2.24. The actual volume inside eachcavity was estimated by measuring the dimensions of the loudspeaker boxes. The volumeare however the same for each cavity.

Qk =

∣

∣

∣

∣

V0,k

c2ρ0iωpk

∣

∣

∣

∣

(2.24)

where,

ρ0 = 1.204 · 10−3 kg/m3

c = 343 m/s

V0,k = 0.007 m3

Inserting the values above into equation 2.24 gives the following relationship betweenmeasured pressure and volume velocity,

Qk = 4.9418 · 10−5· ω · pk (2.25)

which is being used upon conversion.

2.4 Results

This chapter presents the results of the calibration measurements. All plots are smoothedby a moving average filter with a width of 16 samples for clarity. For all individualloudspeakers’ sound power levels, see appendix D.

To test the loudspeakers monopole behaviour and the break frequency, the sphere mea-surements was projected on a virtual sphere, see appendix E. For all different frequencyranges, the value for each position is the mean of the sound pressure level measured forthat microphone. As mentioned in section 2.1.1, the break frequency should be muchlower than 330 Hz.

It can be said that the break frequency for pressure chamber behaviour is indicated bythe cavity pressure measurements in section 2.4.2.

2.4.1 Sound Power Levels

Most of the loudspeakers output very similar sound power levels, which can be seen infigure 2.3. The obvious difference is for elements 27 and 29, which are located on the backside of the ENS (see appendix D). Even though all elements have a cavity of the samevolume, these have quite different dimensions compared to the other elements. For bottommounted elements, a more drastic difference would have been expected. Overall betweenthe loudspeakers, there a maximum standard deviation of about 1.8 dB below 1000 Hz.The SWL measurements can be found in appendix D.

2.4. RESULTS 13

Figure 2.3: Full spectrum of the standard deviation of sound power levels for the differentloudspeakers 01 – 29.

2.4.2 Loudspeaker Cavity Pressure

The standard deviation between the different loudspeaker cavities are about 1 dB below500 Hz. This could be expected mainly because they have the same volume, even thoughthe different cavities differ in size dimensions. Above 500 Hz, the pressure chamber crite-rion falls apart, and the measured pressure is affected by resonances and standing wavesinside the boxes, see figure 2.4.

The cavity measurements can be found in appendix A.

2.4.3 Loudspeaker Volume Velocity

Since the pressure chamber criterion falls apart due to standing waves inside the cavities,the conversion from pressure to volume velocity will be bad above 500 Hz (see figure 2.5).Since the transfer functions are being constructed between the two entities, it doesn’tmatter unless the dimensions of the box are changed while keeping the same volume.

The extracted volume velocities can be found in appendix B.


Figure 2.4: Standard deviation of sound pressure levels inside the loudspeaker cavities 01– 29

Figure 2.5: Standard deviation of the volume velocities inside the loudspeaker cavities 01– 29

2.4. RESULTS 15

2.4.4 Calibration Transfer Functions

Since the calculated volume velocities are small numbers compared to measured soundpower, the calculated transfer functions resembles the volume velocities. See figure 2.6.

This similarity shows the importance of having the correct calibration data for the surfacemicrophones inside the cavities, because a measurement error during runtime of the ENS

could result in high error. This is however indicated by the ENS software, so the user canresolve the issue.

All calculated transfer function can be found in appendix C.

Figure 2.6: Full spectrum of the standard deviation between the constructed transfer func-tions for the loudspeakers 01 – 29.


Chapter 3

Software Development

3.1 Theory

To design an effective interface for the user, there are some key factors that has to betaken into consideration. Bruce Tognazzini explains these principles in his work ”FirstPrinciples of Interaction Design“ [8] namely,

• Autonomy

• Consistency

• Efficiency of the User

• Explorable Interfaces

• Latency Reduction

• Learnability

• Protect Users’ Work

• Readability

• Visible Navigation

The interface needs to be visually forgiving, with a layout that is easy to grasp andunderstand. The internal workings of the application doesn’t need to be explained to theuser, and the settings should be saved at a continuous basis. The term ”Less is More“ isoften an appropriate approach.

Autonomy describes the automatic behaviour of the interface, such as displaying a con-tinuous feed of status information. The environment shouldn’t be too autonomus, whichcould make the user feel that he/she isn’t in charge. However, the interface shouldn’t becompletely manual, as it might feel hazardous or very hard to operate.

Another important aspect is to design the interface with Consistency, with respect to but-ton positions, fields, status information etc. There are some different levels of consistency[8],

17

18 CHAPTER 3. SOFTWARE DEVELOPMENT

1. User Behaviour Interpretation

2. Invisible Structures

3. Small Visible Structures

4. The overall ”look” of a single application or service

5. A suite of products

6. In-house consistency

7. Platform-consistency

The most important consistency is the User Behaviour Interpretation, such as makingsure that shortcut keys maintain their functions throughout the application. InvisibleStructures, and Small Visible Structures comes next, describing hidden windows thatmight occur, buttons, icons, input text boxes etc.

User Efficiency is about keeping the user occupied. The application isn’t efficient if theuser has to wait for processes etc. to finish. The interface should try to optimise thelatancy between user calls. The system architecture is often the key to high efficiency.The principle of Latency Reduction is also linked to efficiency. It is important to reducethe latency experienced by the user, and to always try to optimise speed of the application.Multi-threading is useful if it is available.

A key to the autonomy principle is Explorable Interfaces. It can be compared to a naturallandscape, with roads and landmarks. It’s also important to have some sort of ”home“ forthe user to fall back to. The principle of explorable interfaces also suggests that actionsthat can be performed by the user should be reversible, and to always allow an ”undo“action, so that the user can explore without fearing the effects of the different actions.

The optimal application would have no learning curve, so that any user can start usingthe program and know it all right away. This is in practice not achievable, and there arealways a trade-off between usability and learnability.

The principle of Protecting Users’ Work is important, because data loss can be very criti-cal. The users work should be protected from user error or any program error. Automaticsaving is an option to consider.

3.2 Database for Project Settings

Matlab provides a structure class, similar to structures in other languages such as C/C++.This was chosen over the option of creating an own database class, because of the defaultpublic accessibility and its ease of use. The structure is called Setup, and contains fieldsfor all settings as well as data vectors.

The root of Setup contains several substructures which defines sound groups, loudspeakerassignments, triggers, GUI settings, calibration data and simulation settings. Whichspecific parameters to save was decided as the GUI was developed and programmed, tosuit the needs of the program.

For storing all sound groups and loudspeaker assignment settings, the Setup structure isdynamic, and can change its size depending on the number of groups, references etc.

3.3. MAIN GUI WINDOW 19

The Simulation parameters that are saved are Time, Number of Runs, Bandwidth, Sam-pling Frequency, Number of Bits and IntMax. Time specifies the simulation time in secondsand each simulation run is repeated until the Number of Runs are reached. Sampling Fre-quency is a parameter that is determined by the bandwidth, and is simply calculated usingthe Nyquist sampling theorem,

Fs = 2Bw (3.1)

The Number of Bits are hardware specific, and should be set to match the current DataAcquisition Processors used. For this thesis, the number of bits are always 16, and becauseall numbers are signed, the IntMax (integer maximum) gets a value of 32767. This valuecan be calculated by,

IntMax = 2NB−SB− 1 (3.2)

where,

NB = Number of Bits for Value

SB = Number of Bits used for Sign

To make some parts of the code more efficient, a substructure called Count exists. Itcontains the number of Groups, References, Speakers and Sides. The variables are updatedwhen the information changes, for example if the user creates another sound group.

There is an array of Group substructures, that is used to store information about a soundgroup. Each sound group has a Name, a vector containing all Speakers associated with thegroup, a Sound structure which contains the information and sound data for the group.The Group substructure also contains filtering information, stored in the Filter settingsstructure.

To keep the information about each loudspeaker assignment, the substructure Assignis used. It contains values for which Group it belongs to and if the loudspeaker is aReference. There are also information about which loudspeaker is used as a reference forall loudspeakers.

All calibration functions and volume settings are stored in the Calibration substructure.It has three different level values, and they specify the level in decibels, voltage andinteger value respectively. The calibration data is also saved in the Setup structure, andcontains the calibration transfer function, sensitivity for the surface microphone insidethe loudspeaker cavity, an optional value for serial number specification of the surfacemicrophone and cavity volume for each loudspeaker. The measured volume velocity at1 V, from the calibration measurements, is also saved.

3.3 Main GUI Window

The GUI is separated into several components, handling different actions or programs.

The main GUI window is designed to give the operator access to all settings. Because thesettings are split between different sections, the main window has buttons which openssetting categories in new windows. The main GUI window also handles starting andstopping the engine, and displays status messages from the ENS, see figure 3.1.

For the data preparation and run phases, there are some logging textboxes (located atthe middle left section and top right section respectivly), which displays some runtime


Figure 3.1: Overview of the ENS main GUI window.

information. This can give more information about any error that might occur. It is alsoused to display an overview of the complete settings structure. The outputs of the loggingtextboxes can be saved to normal DOS formatted text files.

The main window also displays the estimated output in sound power level for each loud-speaker, compared to the requested levels. There is also a plot of the mean output soundpower level over frequency for all active loudspeakers.

The main settings categories are:

Save/Load This panel has the options to save or load a project.

Check/Import Calibration Used for loading calibration data to the ENS.

Group Settings Contains all settings for all sounds groups.

Assign Groups Contains all loudspeaker assignments to sound groups.

Trigger Settings All trigger settings are accessed by this panel.

Adjust Levels Used to adjust the output levels of each element.

Generate Data This is used to generate the data for all loudspeakers.

3.4. CHECK/IMPORT CALIBRATION 21

3.4 Check/Import Calibration

Figure 3.2: GUI: Import calibration data.

The window called Import Calibration handles the calibration functions. These are im-ported from files, so that the calibration procedure can be performed in a measurementsystem like Lms. Any other measurement system that can measure Autopower Spectrumwith linear amplitude scaling should work, given that the user saves and organises thedata according to appendix G.12. The design of the window can be seen in figure 3.2.

The measurements to be imported should be measured with the ISO standard 3745 [5].The software expects 1 V RMS to be the reference input signal. If a different outputamplitude is used, the measurements has to be modified manually before importing themto the ENS.

The Import function reads a file which contains the folder architechure for the exporteddata. How the file should be structured can be seen in the manual (appendix G.12).

One requirement was to have a calibration sequence, where each loudspeaker and itsmicrophone is enabled one at a time. During this time, the ENS plays a specified sound(for example white noise), and measures the pressure inside the loudspeaker cavity (thefeedback signal), to check the calibration functions. This in a way that the user can easilybe able to verify the working condition of the ENS.

The function Check Calibration enables the user to quickly test each loudspeaker elementand check the volume velocity for the element, and compares this to the saved volumevelocity from the calibration measurements. This operation is performed in a new window(see figure 3.3). The function sends white noise, with an output amplitude of 1 V RMS.A too high difference between the calculated volume velocity and the nominal volumevelocity could indicate calibration, or microphone error.


Figure 3.3: GUI: Check Calibration Window.

3.5 Group Settings

The group settings window was designed for creation and editing of sound groups. Thereare functions to add (duplicate) groups, remove groups, rename groups etc.

There are a few selected sounds to choose from, based on standard sounds in many signalprocessing applications [9],

• Sinus

• Noise

• Sound File

• Sinus Sweep (Chirp)

3.5.1 Sinusoid Settings

For the sinusoid, the adjustable parameters are simply frequency and phase. Frequency isadjusted in Hz, available from 0 to fs/2, where fs is the output sampling frequency for theData Acquisition Processor card and depends on the bandwidth used. The default valuefs = 17361Hz gives a bandwidth of 34722 Hz (Nyquist theorem). The phase is defined indegrees.

Because the scaling in amplitude of the signal is done separately of the generation, theequation for the sinusoid used is simply,

x(t) = sin(

2πft+( π

180φ))

(3.3)

3.5. GROUP SETTINGS 23

Figure 3.4: GUI: Edit Groups Window.

To output a stationary bias voltage, the following settings can therefore be used,

f = 0 Hz

φ = 90 degrees

3.5.2 Sine Sweep Settings

The speed of the sweep depends on the simulation time setting, because the sweep willspan the simulation time. The sweep (chirp) option has two adjustable parameters, startand stop frequencies.

3.5.3 Noise Settings

For noise, there’s only one option, which is what type of noise to generate. The choicesare White/Uniform or Pink.

White noise is defined as having a constant power spectral density function over all fre-quencies [9],

SX,X(ω) = α (3.4)

White noise is provided by Matlab by the gaussian noise command randn(). The seedfor the function is randomized upon ENS software startup.

Pink noise has equal amount of energy in every one-third-octave band, analog to pinklight. This can be generated from white noise by filtering.


3.5.4 Sound File Settings

The sound file has only one option. The path for the .WAV-file. The sound file willautomatically be resampled to the correct sample rate when preparing the data.

It wanted, the sound file can be given a time delay during the assignment phase of thesetup (section 3.7). It can also simply change the sign of the sound (inverting), which ismuch faster.

3.5.5 Filter Settings

There is also an option to filter the sound, and there are four filter types available, low pass,high pass, band pass and band reject. The filter coefficients are based on the Butterworthequations. Fourth order filter examples can be seen in figure 3.5.

0 0.5 1 1.5 2 2.5 30

0.2

0.4

0.6

0.8

1

Normalised Frequency Ω

Ma

gn

itud

e |H

(Ω

)|

Magnitude Response

0 0.5 1 1.5 2 2.5 3−3

−2

−1

0

1

2

3Phase Response

Normalised Frequency Ω

Ph

ase

φ(

Ω)

LowpassHighpassBandstopBandpass

Figure 3.5: 4th order digital filters.

The parameters for each filter are Filter Order, Cutoff Frequency Left and Cutoff FrequencyRight.

The break frequencies are concatenated into a two word vector for low and high cutofffrequencies. For low and high pass filters, only the low frequency value is used.

There is also an option to open another window to display the frequency and phase re-sponse of the filter. This uses the Matlabs freqz() command. It’s a quick way ofverifying that the filter will be functional.

3.6. TRIGGER SETTINGS 25

3.6 Trigger Settings

To enable the user to specify trigger settings, the trigger settings window was designed.The trigger can be set accordingly to the criterion given in section 1.3, however only theoutput trigger was implemented.

For clarity, the Enable buttons are color coded. They will be green for On, and red for Offstatuses. See figure 3.7.

The output trigger is used to send a trigger to an external system upon start and/or stopof the ENS operation. The output trigger level can be set to any value between 0 – Vout,ref .

The Pulse Width specifies the width of the trigger pulse in seconds, and the Post Delaysets a waiting time before the start pulse is sent. This is the time after the ENS has startedplayback. If a pulse period is specified, the output trigger will produce a pulse train withPulse Period seconds between the pulses.

This makes 4 different types of trigger signals that can be produced, which can be seen infigure 3.6.

(a) (b)

(c) (d)

Figure 3.6: Displaying four different types of output trigger signals. Start (a), Stop (b),Start/Stop (c) and Pulse Train (d).


Figure 3.7: GUI: Edit Trigger Settings window.

3.7 Assign Groups

The Assign Groups settings window lets the operator assign each loudspeaker to a soundgroup, created with the Edit Groups window (see section 3.5). Each side of the ENS isedited one at a time.

Each loudspeaker can be given an unique name. The default name for any loudspeakerN is just Speaker N . Every loudspeaker can be turned on/off independently. The defaultnumbering is given by the hardware connections (see figure 3.8). The left side loudspeakershas the numbers [13, 14, 15, 16, 17, 18, 21, 22], in the same order as the right side. Loud-speakers 27 – 29 are located on the back side, and loudspeakers 23 and 24 are the bottom.

Figure 3.8: ENS loudspeaker numbering.

3.8. ADJUST LEVELS 27

There can be Nref references defined, so that 0 ≤ Nref ≤ N , where N = 29 is the totalnumber of speakers on the ENS. Each loudspeaker can then be given a relative phase to aselected reference.

There is also a 3D overview of the ENS which gives a clear overview of the overall soundassignments, see figure 3.9. The view can be rotated to show every side of the ENS (8 pre-defined views and any custom view).

Figure 3.9: GUI for the loudspeaker assignments.

3.8 Adjust Levels

The GUI together with the calibration functions should enable the user to specify anoutput sound power level (SWL). This is done in the Adjust Levels setup window, seefigure 3.10.

The function estimates the correct voltage signal needed from the system in order toachieve the correct SWL. However, mostly due to non-linearity in the power amplifierchain, the error increases with higher SWL. The user in given a warning when the specifiedSWL gives high amplitudes. The best region is for 0 – 3 V output amplitude. The functionto estimate the voltage signal for loudspeaker k is defined as (derived in section 2.1.2),

vout,k =

√

W010LW,k/10

Hc,k

(

c2ρ0iωVk

)

(3.5)

where,

W0 = 10−12 Watt

LW,k = Specified Sound Power Level for loudspeaker k

Hc,k = Calculated Calibration Function for loudspeaker k

c = Speed of Sound (343 m/s)

ρ0 = Density of Air (1.204 · 10−3 kg/m3)

Vk = Loudspeaker Cavity Volume (0.53 m3)


After the SWL is specified, the user can tune the voltage signals. This runs an algorithmwhich is very similar to the Check Calibration (section 3.4) program. It sends white noisewith the estimated voltage signal, and iteratively update the amplitude until the measuredvolume velocity corresponds with the specified SWL. The updating rule is,

V = V + V

Lw, target − Lw, estimated

40(

epocNiterations

)

(3.6)

where Lw, target and Lw, estimated are the calculated mean value of the target and measuredsound power levels for the specified range. V is the output voltage amplitude, epoc is thecurrent iteration number and Niterations is the total number of iterations to run. Theconstant 40 in the updating rule was decided based on trials, and was set so that thechange in output level would not be too high.

To register and calculate estimated sound power level, Adjust Levels uses the functionsdescribed in section 3.9.

If the analysis of the feedback signal would get a mean level below 20 dB, the functionindicates signal error and aborts the adjustment. This also sets the voltage signal to0 V, to prevent unwanted amplitudes. A signal error could for instance mean that nomicrophone is connected to it’s respective input channel on the backend, malfunction inthe ICP amplifier or error in calibration data.

For all calculations, the sound power is specified in Watt, not in dB. Only when displayinggraphs, the sound power is converted to sound power level. This to reduce complexity ofthe calculations and thereby calculation time.

Figure 3.10: GUI to adjust output levels.

3.9. ACQUIRE INPUT SIGNALS 29

3.9 Acquire Input Signals

To read input signals from theData Acquisition Processors, theMatlab command DAPGETM()(short for DAP Get Matrix) is used. The measured signals being read is in integer format,so to convert the signals to a voltage representation, the signals needs to be scaled. TheICP amplifier has a gain of 10, which might give problems if the signal is more than 1 V inamplitude, because the maximum voltage swing of the Data Acquisition Processor analoginputs is ±10 V.

The scaling procedure is defined as,

Vvoltage = Vref

(

Vinteger

Iref

)

(3.7)

where Vref = 10, Vinteger is the registered input signal in 16 bit integer format, Vvoltage isthe scaled voltage signal between ±10 V and the absolute maximum value that the inputinteger signal can hold is defined as Iref = 32767. (See section 3.2).

3.10 Data Preparation

The data preparation can be divided into several phases,

1. Data Generation

2. Phase Shift

3. Filtering

4. Scaling

5. Organisation

6. Generate DAPL 2000

The first step, Data Generation, creates the raw sound vectors for each active sound type.Some sounds, such as sinusoid, is given the Phase Shift during creation, while a soundgroup using a .WAV-file introduces the phase shift at a separate step.

After the raw sound vectors are created, each sound is filtered if the sound group has anyactive filter, and then scaled to the appropriate output amplitude.

The last step generates DAPL 20001 code, for uploading to the Data Acquisition Proces-sors, making sure that the DAP cards will be properly configured. The Data AcquisitionProcessors have their own programming language, which is interpreted by its on boardprocessors. The language is called DAPL 2000, and is fairly similar to the Analog DevicesDSP Assembler language. The main difference is that DAPL 2000 uses pipes, to definehow the data is moved between processes.

1DAPL 2000 is a assembler scripting language for Data Acquisition Processors, created by MicrostarLaboratories.


3.10.1 Data Generation

Given the setup playback time tp and sampling frequency fs, a time vector is constructedby,

TS =

[

0,1

fs, 2

1

fs, 3

1

fs, . . . , tp

]

(3.8)

For every active loudspeaker, its sound is generated individually, but if the current loud-speaker is using a sound that is already present in memory, it is not regenerated to reduceworkload. For example sound files are only loaded from the hard drive the first time, andthen being kept in memory during the preparation phase.

When preparing the sound file, the function first checks if the file exists. If if does notexist, the function aborts, leaving a zero-vector as output. Otherwise it will load the soundfile into the sound group data vector.

The sound file is also converted to mono sound, by taking the mean between the channels.The sound is also re-sampled to match fs of the Data Acquisition Processor. This is doneby using Matlabs resample() function, and using fs/k. It is then downsampled bytaking every kth sample. This is to preserve high frequency information, which get lost ifusing resample() with only fs, due to the built in filtering. The constant k can be setin the program, and is 2 by default, and was decided by trial for normal sound files with44100 Hz sample rate.

If the sound is shorter than the simulation time, the sound will be repeated by using acrossfader function. The crossfader uses hanning window of 1024 samples as a base (512for fade in, and 512 for fade out). This is done to get rid of transients, which could beharmful to the system. A comparison between a standard repetition of a soundfile and acrossfaded version can be seen in figure 3.11. Another way of removing transients is touse a low pass filter with the cutoff frequency equal to the bandwidth frequency.

0 500 1000 1500 2000 2500 3000 3500 4000

−0.5

−0.4

−0.3

−0.2

−0.1

0

0.1

0.2

0.3

0.4

Crossing at ± 2000 samples

WindowedRepeated

Figure 3.11: Repeating of the wavefile. Displaying and a zoomed in segment over therepeating part which shows how the transient is limited.

3.10. DATA PREPARATION 31

3.10.2 Filtering

The filters are created from the Butterworth equations, using a Finite Impulse Response(FIR). Matlab calculates the coefficients from given break frequencies and filter order.The signal is then filtered by Matlab’s filter() function, using the transfer function,represented in Z-domain,

H(z) =B(z)

A(z)=

b(1) + b(2)z−1 + . . .+ b(n+ 1)z−n

1 + a(2)z−1 + . . .+ a(n+ 1)z−n(3.9)

The signal is normalised before and after filtering.

3.10.3 Data Organisation

Before running sound through the ENS, sound data vectors need to be prepared. To sendsound data to more than one output on the Data Acquisition Processor card, the samplesneed to be organised in a certain manner. The mathematical description for the structureis,

S = [ s1p0, s1p1, . . . , s1pNk, . . . , sNs

p0, sNsp1, . . . , sNs

pNk] (3.10)

where,

si = The i:th sample for the sound

pk = The k:th port to send the data to

Ns = Number of samples in the sound vector

Nk = Number of active output ports

A simplified example of the data organisation can be seen in table 3.1, where Nv is thetotal number of samples in the finished organised sound data vector.

Table 3.1: Simplified example of the sample organisation for two active ports (0 and 3)receiving the same sound data.

Resulting Vector Index Sound Sample Active Output Port

1 1 02 1 33 2 04 2 3...

......

Nv − 3 Ns − 1 0Nv − 2 Ns − 1 3Nv − 1 Ns 0Nv Ns 3

The prepared sound is added to the organised sound data vector mentioned above oneat a time. This is done by upsampling the sound vector by adding Nk − 1 zeros. Theupsampling is done with an offset given by which index the specific loudspeaker has in thePortvec vector (see section 3.10.4).

These vectors are then saved as raw data to the files SendVec0.dat and SendVec1.dat, usedfor streaming the sound to the Data Acquisition Processors.


3.10.4 DAPL 2000 Configuration Files

To configure the DAP 5016a cards, .DAP files are being uploaded to the system. Thesecontain a code segment of the scripting language DAPL 2000 and configures input andoutput configurations of the cards. An example DAPL 2000 code segment can be found inlisting 3.1.

It uses the DAP 5016a module TWOWRITE() (if the module is non existent on the DataAcquisition Processor, it is loaded during DAP initialisation) to send data to the analogexpansion boards MSXB076.

The first argument to this function is a pipe to where the data is. In this case the $bininpipe is being used. The second argument is a vector containing all ports that should beused (called Portvec). In this case, all 24 ports are active. The third argument is forselecting which output configuration to use. In this case OP0, which is defined using theODEF (Output Definition) command.

This output is configured to be used as a master (or slave), using the ports for B0 ad-dressing. If the playback of the sounds is less than 10 seconds, the process will buffer1666656 samples before starting playback. If the playback time is larger than 10 seconds,the sounds are streamed from files on the hard drive, and a buffer isn’t needed.

TIME will set the output (or input) update speed for the card, and is hardware dependent.It can be set by specifying a bandwidth when preparing the sound data. The timing alsocontrols which sampling frequency is needed in Matlab when generating the sounds andresampling audio files.

The sampling frequency can be determined using equation,

fs =

(

1

TIME

)

·1

2 · N(3.11)

where N is the number of active ports in portvec and the parameter TIME is the analogupdate time set on the Data Acquisition Processor cards.

In the example above, this would become,

fs =

(

1

1.2 · 10−6

)

·1

2 · 24≈ 17.361 kHz (3.12)

The fastest sampling frequency that can be achieved, by only running one output is,

fs =1

2 · 1.2 · 10−6≈ 416.67 kHz (3.13)

It is, however, more convenient to always use the same sampling frequency.

3.11. COMMENTS 33

Listing 3.1: Example of a DAPL 2000 configuration file

1 RESET

2 OUTPORT 0..23 TYPE=1

3 RESET

4

5 PDEF BMASTER

6 TWOWRITE($binin ,(0..23) ,OP0)

7 MERGE(IP (0..15) ,$binout)

8 END

9

10 ODEF CMASTER 1

11 MASTER

12 SET OP0 B0

13 OUTPUTWAIT 833424

14 TIME 1.2

15 END

16

17 IDEF INMASTER

18 CHANNELS 16

19 SET IP0 S0 1

20 SET IP1 S1 1

21 SET IP2 S2 1

22 SET IP3 S3 1

23 SET IP4 S4 1

24 SET IP5 S5 1

25 SET IP6 S6 1

26 SET IP7 S7 1

27 SET IP8 S8 1

28 SET IP9 S9 1

29 SET IP10 S10 1

30 SET IP11 S11 1

31 SET IP12 S12 1

32 SET IP13 S13 1

33 SET IP14 S14 1

34 SET IP15 S15 1

35 TIME 1.25

36 END

3.11 Comments

The GUI gives the user access to all important parameters for the application. Some partsof the application are quite slow, especially functions that draw/plot data. The solutionwas to introduce checkboxes, so that the user can disable some of the plots.

When generating very large sound data for streaming, there is a limit in size due to thememory usage by the functions. This is not very good, but it could be avoided if thefiles containing the data was written ”On the Fly“ while preparing the data instead ofallocating memory before writing. Due to the lack of time for optimising the program,this was not implemented. At the moment, the only way to produce the largest sounddata is to reduce the bandwidth parameter to the minimum.


Chapter 4

Discussion

Matlabs GUIDE, used when creating Graphical User Interfaces, is an easy to use tool.It provides a drag and drop creation workflow, useful for designing the layout of a GUI. Aprogrammable approach is available, but it can be hard to foresee how the GUI will looklike when running the program.

The GUI was briefly tested, and the users could with help from the manual set up theENS with an engine sound. Based on the theory in section 3.1, the GUI can be consideredgood. There are however no Undo option at the moment, which the user needs to beaware of. Instead of using separate windows for the GUI, different tabs could have beenused instead for added clarity. However, since Matlab don’t natively support tabs, thewindowed approach was used instead. By using native functions, the program will runfaster.

The Microstar Technologies Data Acquisition Processors are very reliable in the sensewhen you define a process, it will perform the same all the time. It was however verytricky to get the process definition to work as wanted. The main problem was to set thesampling frequency of the DAPs. When understanding how the timing parameters and howthe TWOWRITE() module works, it became clear.

Due to the nature of the timing and the output behviour, the decision was to always havethe same sampling frequency, thus always having all output ports active. The outputsare however set to 0 V when not active. The timing for the output was decided to be avariable, set by a bandwidth parameter.

The calibration measurements used when adjusting the output levels works. However, dueto some non-linearity in the signal chain, it is not very easy to estimate the correct outputvoltage level. The non-linearities is most probably in the power amplifier stage.

To evaluate how the ENS works, some recorded sound signals from an anechoic enginetest cell was feed to the engine. It was only evaluated by ear, so no measurements wasdone. By judgment of the quick test, the ENS seems to be able to produce realistic enginesounds, and can be used to evaluate noise encapsulation or perform drive by tests etc. Itis however recommended that further evaluation work should be performed, and that theoutput sound power levels are correct (as specified in the software).

35

36 CHAPTER 4. DISCUSSION

Chapter 5

Future Work

There are a few pointers and suggestions on continued work,

• Improve data generation latency

• Evaluate the ENS by engine modelling, or pass-by test

• Expand the toolset in the program

• Improve overall latency and memory usage

• Implement input triggering

• Improve output trigger, so that pre/post functionality exists

The most critical work to be done if long sounds should be played (over 120 seconds), isto improve the data generation phase. As suggested in section 3.11, the function couldbe rewritten so that hard drive files would be generated directly, instead of allocatingmemory. This would allow for very long vectors to be stored, and then just streamed tothe Data Acquisition Processors.

Since the ENS was only evaluated by ear, future work on the ENS should include a measuredevaluation. As Scania suggested for the scope of the thesis, the evaluation could be doneby modelling a diesel combustion engine or performing a drive-by test using previouslyrecorded engine noise sound data.

The ENS GUI is quite slow when updating plots. The code could be optimised by onlyupdating what’s needed, and not the complete window. Another approach could be tohave options to disable graphical feedback.

37

38 CHAPTER 5. FUTURE WORK

Bibliography

[1] Rieter (www.rieter.com), “The rieter automotive systems Engine Noise Simulator ENSas controllable surrogate engine-gearbox unit for nvh-investigations.”

[2] J March, G Strong and S Gregory, “Achieving diesel vehicle appeal part 1: Vehiclenvh perspective,” January 2005.

[3] “Data acquisition (daq).” http://www.mstarlabs.com, July 2010.

[4] “Mathworks - Matlab and simulink for technical computing.”http://www.mathworks.com, July 2010.

[5] ISO 3745:2003(E), “Acoustics – determination of sound power levels of noise sourcesusing sound pressure – precision methods for anechoic and hemi-anechoic rooms,”December 2003.

[6] H Boden, U Carlsson, R Glav, H P Wallin and M Abom, Ljud och Vibrationer.Stockholm, Norsteds Tryckeri AB: Marcus WallenbergLaboratoriumet, 2001.

[7] M. Kleiner, Audioteknik och Akustik. Goteborg: Chalmers Tekniska Hogskola, Insti-tutionen for teknisk akustik, 2000.

[8] “First principles of interaction design.” http://www.asktog.com/basics/firstPrinciples.html,July 2010.

[9] H Stark, J W. Woods, Probability and Random Processes with Applications to SignalProcessing. New Jersey: Prentice Hall, 3rd ed., 2002.

39

40 BIBLIOGRAPHY

Appendix A

Loudspeaker Cavity Pressure

For all loudspeakers located on the top (speakers 1 – 6), figure A.1, the measured pressureis very similar up to almost 4000 Hz. There is a clear resonance peak at 1000 Hz and ananti-resonance at 1400 Hz.

Figure A.1: Full spectrum of the measured sound pressure levels for Loudspeakers locatedon the top.

1

2 APPENDIX A. LOUDSPEAKER CAVITY PRESSURE

The measurements differ a bit for the speakers located on the bottom (speakers 23 and 24),figure A.2, . This is probably due to different cavity dimensions. The pressure chambercriterion is valid for low frequencies, and given that the two cavities has almost the savevolume, the similarity below 500 Hz is expected. The first peak for element 23 is locatedat 1120 Hz while it is located at 1285 Hz for element 24. For these speakers, compared tothe top speakers, the pressure chamber behaviour breaks down at about the same location,800 Hz.

Figure A.2: Full spectrum of the measured sound pressure levels for Loudspeakers locatedon the bottom.

For the left and right side, figures A.3 and A.4, most of the elements are very similar.The speakers that stand out are the ones located closest to the bottom of the ENS. Thisis probably due to different dimensions compared to the others. All speakers has a clearresonance peak at about 1230 Hz.

The front speakers are very similar up to 3000 Hz. The resonance peak at about 1500–1700Hz is not as clear as for the other speakers and the anti-resonance that occurs before, atabout 1000 Hz. These speakers can be seen in figures A.5.

There is a difference between the loudspeaker located in the middle of the back side ofthe ENS. This has different dimensions compared to it’s neighbours so it can be expected.The first anti-resonance is somewhat surprising, as it occurs fairly early in the measuredbandwidth. They are located at 650 Hz and 800 Hz respectively. See figures A.6.

3

Figure A.3: Full spectrum of the measured sound pressure levels for Loudspeakers locatedon the left.

Figure A.4: Full spectrum of the measured sound pressure levels for Loudspeakers locatedon the right.

4 APPENDIX A. LOUDSPEAKER CAVITY PRESSURE

Figure A.5: Full spectrum of the measured sound pressure levels for Loudspeakers locatedon the front.

Figure A.6: Full spectrum of the measured sound pressure levels for Loudspeakers locatedon the back.

Appendix B

Loudspeaker Volume Velocity

For all loudspeakers located on the top (speakers 1 – 6), figure B.1, the volume velocitiesare very similar. Based on the spectrum limited plots, the volume velocity theorem basedon pressure chamber behaviour seems to break down at about 800 Hz.

For the left and right side, figures B.3 and B.4, below 500 Hz, the similarity between themeasured pressure and volume velocity is very good. The pressure chamber behaviourseems to dissolve at about 500 Hz.

The front speakers can be seen in figures B.5. For these speakers, the assumption thatpressure chamber behaviour exist seem to hold up to 1000 Hz.

For the speakers located on the back side, the peaks located at 650 Hz and 800 Hz respec-tively (see figures B.6), indicates that the pressure chamber assumption is only valid for alower span for these speakers, such as 500 Hz.

Figure B.1: Full spectrum of the calculated volume velocity for Loudspeakers located on thetop.

1

2 APPENDIX B. LOUDSPEAKER VOLUME VELOCITY

Figure B.2: Full spectrum of the calculated volume velocity for Loudspeakers located on thebottom.

Figure B.3: Full spectrum of the calculated volume velocity for Loudspeakers located on theleft.

3

Figure B.4: Full spectrum of the calculated volume velocity for Loudspeakers located on theright.

Figure B.5: Full spectrum of the calculated volume velocity for Loudspeakers located on thefront.

4 APPENDIX B. LOUDSPEAKER VOLUME VELOCITY

Figure B.6: Full spectrum of the calculated volume velocity for Loudspeakers located on theback.

Appendix C

Calculated Calibration Functions

For the speakers located on the top of the ENS, the calibration functions can be seen infigure C.1. As mentioned earlier, the transfer functions are very similar to the volume ve-locities and can be expected. Also the bottom speakers are very similar to the top speakersas can be seen in figure C.2. There are some differences between the two bottom speakershowever, which mentioned when presenting the measurements of the sound pressures, isprobably due to dimension differences of the loudspeaker cavities.

The transfer functions for the left and right side of the engine (figures C.3 and C.4), thepressure chamber behaviour breaks down early, indicated by the early resonances andanti-resonances.

The front and rear speakers (figures C.5 and C.6) display a different calibration functioncurve than for the other loudspeakers, especially for the limited spectrum between 125Hz and 1000 Hz. It is a more constant increasing slope until the first resonance or anti-resonance.

31.5 63 125 250 500 1000 2000 4000 8192

102.32

102.33

102.34

102.35

102.36

102.37

102.38

102.39

102.4

Hz

W/m

3/s

Speakers on Top Side

Element 01Element 02Element 03Element 04Element 05Element 06

Figure C.1: Full spectrum of the calculated transfer functions Hc,k for loudspeakers locatedon the top.

1

2 APPENDIX C. CALCULATED CALIBRATION FUNCTIONS

31.5 63 125 250 500 1000 2000 4000 8192

102.32

102.33

102.34

102.35

102.36

102.37

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102.39

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Element 23Element 24

Figure C.2: Full spectrum of the calculated transfer functions Hc,k for loudspeakers locatedon the bottom.

31.5 63 125 250 500 1000 2000 4000 8192

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102.33

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Element 13Element 14Element 15Element 16Element 17Element 18Element 21Element 22

Figure C.3: Full spectrum of the calculated transfer functions Hc,k for loudspeakers locatedon the left.

3

31.5 63 125 250 500 1000 2000 4000 8192

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Element 07Element 08Element 09Element 10Element 11Element 12Element 19Element 20

Figure C.4: Full spectrum of the calculated transfer functions Hc,k for loudspeakers locatedon the right.

31.5 63 125 250 500 1000 2000 4000 8192

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102.32

102.33

102.34

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102.36

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Element 25Element 26

Figure C.5: Full spectrum of the calculated transfer functions Hc,k for loudspeakers locatedon the front.

4 APPENDIX C. CALCULATED CALIBRATION FUNCTIONS

31.5 63 125 250 500 1000 2000 4000 8192

102.32

102.33

102.34

102.35

102.36

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102.38

102.39

Hz

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Element 27Element 28Element 29

Figure C.6: Full spectrum of the calculated transfer functions Hc,k for loudspeakers locatedon the back.

Appendix D

Measured Sound Power Levels

31.5 63 125 250 500 1000 2000 400040

45

50

55

60

65

70Speaker 1 with Std Limits

dB

Hz

Figure D.1: Measured sound power levels for loudspeaker 1.

1

2 APPENDIX D. MEASURED SOUND POWER LEVELS

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Appendix E

Loudspeaker Directivity Test

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Figure E.1: Directivity test for speaker number 4, located on the top of the ENS.

1

2 APPENDIX E. LOUDSPEAKER DIRECTIVITY TEST

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Figure E.2: Directivity test for speaker number 24, located at the bottom section.

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Figure E.3: Directivity test for speaker number 10, one of the loudspeakers on the rightside.

3

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Figure E.5: Directivity test for speaker number 26, the bottom front speaker.

4 APPENDIX E. LOUDSPEAKER DIRECTIVITY TEST

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Figure E.7: Directivity test for speaker number 29, on the rear left side.

Appendix F

Measurement Instrumentation

Table F.1: ENS instrument specifications

Microphone Name Type SN Sensitivity Manufacturer

1 Reference 01 40PS 90732 4.73 G.R.A.S





























1

2 APPENDIX F. MEASUREMENT INSTRUMENTATION

Table F.2: Sphere instrument specifications

Microphone Name Type SN Manufacturer

1 POWER 1 46AE 109834 G.R.A.S

2 POWER 2 46AE 119335 G.R.A.S

3 POWER 3 46AE 119336 G.R.A.S

4 POWER 4 46AE 119337 G.R.A.S

5 POWER 5 46AE 119338 G.R.A.S

6 POWER 6 46AE 119339 G.R.A.S

7 POWER 7 46AE 119340 G.R.A.S

8 POWER 8 46AE 119341 G.R.A.S

9 POWER 9 46AE 119342 G.R.A.S

10 POWER 10 46AE 119343 G.R.A.S

11 POWER 11 46AE 67718 G.R.A.S

12 POWER 12 46AE 67719 G.R.A.S

13 POWER 13 46AE 67720 G.R.A.S

14 POWER 14 46AE 67721 G.R.A.S

15 POWER 15 46AE 67722 G.R.A.S

16 POWER 16 46AE 67723 G.R.A.S

17 POWER 17 46AE 67724 G.R.A.S

18 POWER 18 46AE 67725 G.R.A.S

19 POWER 19 46AE 67726 G.R.A.S

20 POWER 20 46AE 67727 G.R.A.S

Appendix G

Manual

1

2 APPENDIX G. MANUAL

G.1 Quick Start

The workflow for the ENS software is divided into several steps,

1. Connect all cables and components

2. Start the computer

3. Start the power amplifiers and ICP amplifiers

4. Load or create a new project. Give the project an appropriate name

5. Check, or import, calibration data

6. Check, edit, remove and create sound groups and filters

7. Assign each active loudspeaker with a sound group

8. Setup the trigger as wanted

9. Adjust the levels of each loudspeaker

10. Generate the data to be used

11. Verify the data

12. Run the ENS

The Quick Start chapter will guide you through the process. If you want more specificinformation about each section, please refer to it’s own chapter in the manual.

G.1. QUICK START 3

Figure G.1: Hardware Rack.


G.1.1 Connection of the Cables

The routing used by the ENS between the input backplane and the ICP amplifiers can befound in table G.1. The routing layout should be pretty self explanatory by the table. Forexample, BNC connection 3B on the left backplane should be connected to the 10th outputof the top mounted ICP amplifier. BNC connection 4D on the right backplane should beconnected to the 16th output of the bottom mounted ICP amplifier.

Ports 5D on the left and right backplane are used as input triggers. Output trigger isrecieved from port 24 on the slave Data Acquisition Processor expansion MSXB 076 card.

The feedback signals are then connected to the ICP inputs, labelled 1 – 32. Only the first29 are used, however.

Table G.1: Backplane to ICP Amplifier Routing

Backplane Connections ICP Outputs

Left BackplaneDAP 5016a

Master

1A 1B 1C 1D

2A 2B 2C 2D

3A 3B 3C 3D

4A 4B 4C 4D

5A 5B 5C 5D

Connected toTop ICPAmp

⇒

1 2 3 4

5 6 7 8

9 10 11 12

13 14 15 16

-- -- -- IT

Right BackplaneDAP 5016a

Slave

1A 1B 1C 1D

2A 2B 2C 2D

3A 3B 3C 3D

4A 4B 4C 4D

5A 5B 5C 5D

Connected toBottom ICPAmp

⇒

1 2 3 4

5 6 7 8

9 10 11 12

13 14 15 16

-- -- -- --

Figure G.2: Unconnected ICP amplifiers

G.1. QUICK START 5

Figure G.3: Connected ICP amplifiers and feedback signals.

Figure G.4: Connected feedback signals.


The loudspeaker elements are connected by the six large round connectors. The connec-tions is labelled at the backside of the rack, and can also be found here in table G.2.

Table G.2: Loudspeaker connections at the back side of the rack.

Top Row: 19 – 24 13 – 18 7 – 12 1 – 6

Bottom Row: 25 – 26 27 – 29

Figure G.5: Connected loudspeakers.

G.1. QUICK START 7

G.1.2 Starting the Hardware

To start the hardware, power on the computer first. Then the power amplifiers. Thereverse order applies upon shutdown. This is to prevent undetermined output from theData Acquisition Processors when there is no power applied to them.

In general,

1. Start Computer

2. Start Power Amplifiers and ICP Amplifiers

3. Run the ENS

4. Power down the Power Amplifiers

5. Shutdown the Computer

G.1.3 Starting the ENS Software

Figure G.6: Loading Screen. This is automatically displayed when the ENS software isstarting up.


G.1.4 Running Default Settings

The ENS comes with a predefined project. This is automatically opened at startup. Tocheck the calibration status of the ENS, using this default settings, click on the CheckCalibration button. Make sure that the power amplifiers are on, and that all feedbackmicrophones are connected to the ICP amplifier, and that the ICP amplifier is also turnedon.

The default checking time for each element is 1 second, but increasing the value will givea more accurate check. Edit this value if wanted, then press the Check button to run thealgorithm.

Each loudspeaker element is given a signal of 1 V amplitude white (Gaussian) noise. Thepressure inside the loudspeaker cavity is measured, and converted to volume velocity.The table will be updated after each element, displaying if the measurement correspondswell with the calibration values. More information about this procedure can be found insection G.4. After checking, you can close the window.

The default project has a sinusoid sound group assigned to all loudspeakers on the ENS.To be able to run the engine, this data needs to be generated and sorted in a special way.Click on the Generate Data button to create the vectors. The default running time is 4seconds, and only one run in sequence. Change this before generating data, if wanted.

After data generation and sorting, press the Run Engine button to produce sound.

The default settings uses no triggers, so the user can quickly check that all loudspeakersproduces sound without needing to wait for triggers.

G.2. MAIN GUI WINDOW 9

G.2 Main GUI Window

The Main GUI Window is the first thing you see after the loading screen disappears.Since there are many settings to adjust on the ENS, the Graphical User Interface is splitinto several windows.

From the Main GUI Window, you can access these different windows and options neededto configure and running the ENS.

The Main GUI Window has panels for,

Save/Load Settings Handles save, load and project name, data and description.

Calibration Shortcuts for import and check calibration windows.

Setup Handles the settings for the sounds, trigger and output levels.

Mean Estimated Output A graph displaying the mean value of the estimated SoundPower Level output.

Feedback Status The mean Sound Power Level of each loudspeaker (between the spec-ified frequency lines) are indicated here.

Run This panel handles the start and stop of the ENS, and has a logging window whichdisplays various things. Refer to chapter G.10 at page 40 for information aboutrunning a simulation.

Data Preparation Data preparation and displaying the data is handled by this panel.It also has a logging window so you can get feedback about what is happening. Seechapter G.9 at page 37 for more information.


The Main GUI Window has a menu bar, altough most of the options in the menu bar isalso available as buttons on the Main Window. The following menu bar items exists,

File → New Project

New Project from Template...

Save Project

Save Project as...

Restart Software

Quit

Settings → Group Settings...

Assign Groups...

Trigger Settings...

Adjust Levels...

Calibration → Import Calibration...

Check Calibration...

Data Preparation → Generate Data

Save Preparation Log

Clear Preparation Log

Preview → Preview Data...

Preview DAPL Configurations...

Internal ENS Settings → Show

Edit...

Help → About...

Open Manual (PDF)

Table G.3: Main Window menu bar structure.

G.2.1 Creating a New Project

To create a new project, click on the File menu in the menubar, then either New Project

or New Project from Template.... If this is the first time running the ENS, New Project

is most likely the option to use.


New Project

The New Project option sets the complete ENS to default state. The default settings forall options can be found in appendix G.11, on page 43.

New Project from Template. . .

The New Project from Template... option let you create a new project by inheritingthe settings from a previously saved project. This can be useful if you have created a basicproject that will be used as a base for many new projects in the future. When selectingthis option, a dialog appears where you can select the template project file.

Note that any previously saved project can be a template!

G.2.2 Saving the Project

To save the project, you can press the Save As. . . or Save button. If you have createda new project, and is saving for the first time, both buttons do the same thing, namelypresents you with the Save As. . . prompt,

G.2.3 Importing and Checking Calibration

This section gives some quick information about the Calibration panel.


Import Calibration

To import calibration data to the project, press the Import Calibration button. Thisopenes a new window which handles the importing. More about this in chapter G.3, onpage 16.

You can also open this window using the menu bar by selecting Calibration → ImportCalibration.

Check Calibration

To ensure that the level settings will be accurate, you need to regurarily check the cali-bration status of the ENS. This is done by opening a new window by pressing the CheckCalibration button in the Calibration panel. The window can also be opened using themenu bar by selecting Calibration → Check Calibration.

More about this in chapter G.4, on page 17.

G.2.4 Setup

This section gives some quick information about the Setup panel.

Group Settings

The add, edit or remove sound groups, you need to open the Edit Group Settings window.This is done by either pressing the Group Settings button, or using the menu bar byselecting Settings → Group Settings.

More about this window in chapter G.5, on page 21.

Assign Groups

For a loudspeaker to be able to produce sound, the loudspeaker needs to be assigned toa sound group. This is done in the Assign Groups window. Open the Assign Groupswindow by clicking the Assign Groups button, or by using the menubar.



Trigger Settings

The input and output trigger settings are handled in the Trigger Settings window. Thisis opened by pressing the Trigger Settings button, or by using the menubar.

Adjust Levels

The ENS hardware uses integer format to specify the output level of each loudspeaker.This is why the calibration data is used, so you can specify the output Sound Power Level(SWL) instead.

To specify all loudspeaker output levels, open the Adjust Levels window by clicking onthe Adjust Levels button. This can also be opened using the menubar, by selecting Set-tings → Adjust Levels.


G.2.5 Mean Estimated Output

When running the ENS, some feedback data is acquired. This panel displays a mean valueof all acquired sound power levels for the loudspeakers.

G.2.6 Feedback Status

This panel displays a bar graph och the mean sound power levels produced by each loud-speaker. The mean is calculated between the frequency lines specified in the Adjust Levelswindow.


G.2.7 Help

This panel has shortcuts for the menubar that opens an about window and the manual.

G.2.8 ENS Internal Settings

The ENS Internal Settings contain information about the ENS loudspeaker structure andsome hardware settings. These should not be changed unless you really know what you aredoing! To show the current settings, press the ENS Internal Settings → Show menubaroption. If you want to edit the settings, click on ENS Internal Settings → Edit. . . .

In case you edit the ENS Internal Settings, you will need to restart the software. This canbe done by selecting File → Restart Software in the menubar.


There are some GUI specific settings that you can edit in the window, such as font, sizeand the colors of the input boxes. These can be set freely as preferred. Before you canedit any settings, you need to press the Unlock button twice. Edit any setting, then pressSave Settings to save.

Figure G.7: ENS Internal Settings Window. This is where you can change importantsettings of the ENS.


G.3 Import Calibration Window

Figure G.8: Import Calibration Window.

G.3.1 Import Calibration Data

Before the ENS is fully operational, some calibration data need to be imported. This isdone in the Import Calibration Data window. The function Import Calibration Data loadsa textfile, containing a special file structure. The definition of this file can be found inappendix G.12, on page 46.

The calibration performed by Andre Lundkvist, at 2010-05-06, is already loaded by default(unless a new ”Default.mat“ file is written).

G.4. CHECK CALIBRATION WINDOW 17

G.4 Check Calibration Window

Figure G.9: Check Calibration Window.

To quickly verify the working condition of the ENS, you can use the Check Calibrationfunction. It will play a white noise over the given playback time, for all loudspeakers. Whenplaying, it measures the sound pressure levels inside each loudspeaker cavity, converts itto volume velocity and compares the result to the calibration data.

G.4.1 Select Speakers

To perform a check, first select the speakers you want to check. This is done in theCheck the Following Loudspeakers panel. You can press the Toggle All button to switchall loudspeakers On or Off.


G.4.2 Running the Check

To start the check, specify the measurement length in the box that says Test Length(seconds). After you are happy with the measurement time setting, click on the big greenbutton Check Calibration.

When you start the check, there will be a warning about loud levels. Click Ok to proceed.

While Running. . .

When the check is running, you can abort by pressing the starting button again, this timecalled Running Calibration. . . button.

During the checking, some data is acquired. You will be presented with the time signal,updating as the process is running.

The time signal will be converted to different entities after the measurement time is com-pleted. These are SPL inside the loudspeaker cavity, volume velocity and also the errorbetween the calibration volume velocity and the measured.

G.4. CHECK CALIBRATION WINDOW 19

The Volume Velocity panel displays the measured volume velocity (blue line) and the cal-ibration volume velocity (dotted magenta line). Note that for this manual, the measure-ment is running in Demomode, thus not displaying an actual volume velocity measurement.

Calibration Status Messages

The function will indicate the results in the table, where the status can be,

Ok! The loudspeaker has correct calibration and is working properly.

Warning! Displayed if the maximum squared error between the measured volume velocityand the calibration volume velocity is higher than 0.005, but lower than 0.015,indicating that there are some differences, and should be noted, as the output levelsmight not be totally accurate.

Error! If the maximum squared error is higher than 0.015, there might be a calibrationerror, or it might be that no signal is recieved.

DEMO Mode Displayed if the ENS is running in Demo mode.


The Maximum Squared Error between two vectors X and Y is given by,

ǫ2 = MAX (X[k]− Y [k])2 (G.1)

The error is also displayed over frequency in the Squared Error over Frequency panel.

G.5. EDIT SOUND GROUPS WINDOW 21

G.5 Edit Sound Groups Window

This section discusses the Edit Sound Groups Window. It contains all sound groups thatare created. You can create, remove and reset groups.

There are some different sound sources that can be selected, each with different settings.There are also filtering options.

Figure G.10: Edit Groups Window. The figure here displays the default setting.

G.5.1 Sound Groups & Filters

Sound groups defines what sounds the loudspeakers produce. There can be any numberof sound groups, and they have 4 sound options. These are sinusoidal output, noise(white/gaussian or pink), sinus sweep or soundfile import.

There are also a number of different filters that can be applied in the sound, which arethe four traditional Butterworth filters. The filter order, as well as break frequencies, canbe specified.


G.5. EDIT SOUND GROUPS WINDOW 23

You can display and inspect the filter curve by pressing the Display button. Note thatwhen using No Filter, the button is disabled.

Low Pass High Pass

Band Pass Band Reject

Add Sound Groups

To add a sound group, first select a sound group which has settings you might want tocopy to the new one, then press the Add (Duplicate) button. This will use the settings ofthe selected group, and copy it to a new sound group. This makes it fast to create somedifferent sound groups with similar settings.

Remove Sound Groups

It is very easy to remove a sound group. Simply select the group you want to delete byselecting it in the drop down curtain selector, then clicking the Remove Group button.This will also set all assignments to the sound group to be updated, using the sound groupafter the removed one in the list.


Resetting Sound Groups

If you feel like scraping all settings of the selected sound group, you can either reset thesound group to the default (which is noise, as explainded above), or you can use theDefault menubar.

The menubar lets you specify any default type of sound, Sinusoid, Noise, Soundfile orSweep/Chirp. You can also reset the filter to the default settings, for either Lowpass,Highpass, Bandpass or Bandreject.

G.6. LOUDSPEAKER ASSIGNMENTS WINDOW 25

G.6 Loudspeaker Assignments Window

The Loudspeaker Assignments window has all options needed to associate a loudspeakerwith a sound group.

You can specify a unique name for any loudspeaker, set it to on or off status, select asound group and specify a relative phase to any of the defined references.

Figure G.11: Loudspeaker Assignments Window.

G.6.1 Naming and Activating

Any loudspeaker can have a any kind of name. For instance, the two front speakers couldbe named Front Upper and Front Lower. It is however recommended for the most part touse the standard names.

The Active switch sets the loudspeaker to On or Off. The Set All toggles all loudspeakerson the side that is currently being edited.

G.6.2 Assigning Loudspeakers to Groups

To associate a loudspeaker to a sound group, click on the drop down box below the UpdateGroups button and select the group. In case some groups are missing, you can press theUpdate Groups button to refresh the groups list.


G.6.3 References and Phase

Any loudspeaker can be set as a reference by toggling the Reference checkbox.

After a reference is created, any loudspeaker can have this as a reference by chosing it inthe To Reference drop down box.

If you add more references, or remove a reference, you might need to check all loudspeakerso that the reference settings was not corrupted.

The relative phase sets a phase of the loudspeaker in relation to its reference. If theloudspeaker has no reference, the phase will be in respect to 0.

G.6.4 Assignments Overview

The Assignments Overview panel displays a 3D image of the ENS. The loudspeakers areindicated by colored squares, and the color depends on which sound group is associatedwith the loudspeaker. To turn on the display, click on the Update Overview button.

If the loudspeaker is Off, the color will be close to the color of the surrounding material.

Updating the Overview

You can update the 3D model by clicking on the Update Overview button. This willre-render the model with the updated group assignments.

You can do this automatically when saving an assigned side by checking the AutomaticallyUpdate Overview checkbox. However, updating the overview is time consuming.

G.6. LOUDSPEAKER ASSIGNMENTS WINDOW 27

Changing the View

To view another side of the ENS model, you can use one of the pre-defined views, or defineyour own. The predefined views are,

Overview Top A 3D overview displaying the top and right side of the ENS.

Overview Bottom An overview of the model displaying the bottom and left side of theENS.

Current Side Displays a 2D overview of the current side that is being edited.

To define your own view, you specify the rotation around the z-axis, and the azimuth.Both the rotation and azimuth is defined in degrees. When you have defined the view,you click on the Custom View to change to the specified viewpoint.


G.7 Trigger Settings Window

G.7.1 Input Trigger

The input trigger is used to start and/or stop the ENS operation. The level can be setto any value between 0 – 10 V, and is used as a threshold level for when the trigger isaccepted.

The trigger can be both enabled and disabled. Disabling the input trigger causes the ENSto perform free runs. If the trigger is enabled, the ENS will wait for the trigger for everysimulation run before playback is started.

However, the Input Trigger is not implemented at this moment.

G.7.2 Output Trigger

The output trigger is used to send a trigger to an external system upon start and/or stopof the ENS operation. The output trigger level can be set to any value between 0 – Vout,ref .

The Pulse Width specifies the width of the trigger pulse in seconds, and the Post Delaysets a waiting time before the start pulse is sent. This is the time after the ENS has startedplayback. If a pulse period is specified, the output trigger will produce a pulse train withPulse Period seconds between the pulses.

G.7. TRIGGER SETTINGS WINDOW 29

Type: Start

Type: Stop


Type: Start/Stop

Type: Pulse Train

G.8. ADJUST OUTPUT LEVELS WINDOW 31

G.8 Adjust Output Levels Window

G.8.1 Adjust Levels Overview

The GUI together with the calibration functions enables the user to specify an outputsound power level (SWL). This is done in the Adjust Levels setup window.

The Adjust Levels window gives you the tools to set the output levels of all loudspeakers.The loudspeaker levels can be edited, one side at a time, or be set for all loudspeakers atonce by using the Batch Set option.

The level is set by extracting a mean level value between to frequency lines, which can bespecified. This can be edited for each loudspeaker by changing the Left [Hz] and Right

[Hz] columns in the level table.

G.8.2 Changing the Output Levels

The output levels can be set by the levers or by manually entering a number in the table.Only the output Sound Power Level (SWL) can be set, in decibels. When adjusting thedb SWL, the voltage amplitude and rms values are updated, by using the calibrationfunction.

The levels is set between the frequency lines, specified by Left [Hz] and Right [Hz] columnsin the level overview table.


Default Levels

There is an option to set all, or only the current side, to the default settings. That is,

Sound Power Level 60 dB

Left Frequency Line 250 Hz

Right Frequency Line 1000 Hz

The only way to set loudspeakers to the default level is to use the menubar, by clickingon Default → Current Side or All Loudspeakers.

Batch Set

If you want to set all levels, or the current side’s levels, to a specific value, you can use theBatch Set option. Here you can specify any sound power level to set between a specificfrequency limit.

To set the loudspeakers, adjust the settings and press the Set All or Current Side buttonin the Batch Set Levels panel.


Level Warnings

If the output level will be too low or very high, there will be level warnings. For a verylow level, the integer value will have low resolution, thus indicating a Resolution Warning.If the level would become 0, due to a tuning error, the function will indicate Total Error.

For a voltage level over 3.5 V, the warning will be High Level, and the function alsoindicates a warning for Maximum Level.

Closing the Adjust Levels Window

When closing the window, the level settings are saved for all loudspeakers. If you havenon tuned speakers, you are displayed with a question,

Click Yes to tune all non tuned speakers before closing, No to just keep the currentsettings, or Cancel if you feel you need to perform any changes to the setup.

G.8.3 Tuning of the Levels

The loudspeakers are tuned all at once or one side at a time, by iterating a voltage levelto get the correct dB SWL.


Tuning Options

You can specify the measurement time of a given period, 1, 2, 4, 6, 8 or 10 seconds. Thelonger time, the better results.

There is also an option to adjust the number of iterations to perform for each loudspeaker.A value of about 8 is usually enough to tune the loudspeakers properly. The number ofiterations is defined by a curtain menu, by selecting any of the values 1, 4, 8, 16, 32 or 64.

You can also set a target error limit for the difference in estimated sound power level.This will cause the iterations to stop when the difference is better than the target. If youwant the iteration to never abort, set the target to 0. The Source option specifies whichtype of output sound data should be used.

Updating the Voltage Levels

The function estimates the correct voltage signal needed from the system in order toachieve the correct SWL. However, mostly due to non-linearity in the power amplifierchain, the error increases with higher SWL. The best region is for 0 – 3 V output amplitude.

After the SWL is specified, the user can tune the voltage signals. This runs an algorithmwhich is very similar to the Check Calibration program. It sends white noise with theestimated voltage signal, and iteratively update the amplitude until the measured volumevelocity corresponds with the specified SWL. The updating rule is,

V = V + V

Lw, target − Lw, estimated

40(

epocNiterations

)

(G.2)

where Lw, target and Lw, estimated are the calculated mean value of the target and measuredsound power levels for the specified range. V is the output voltage amplitude, epoc is thecurrent iteration number and Niterations is the total number of iterations to run.


For all calculations, the sound power is specified in Watt, not in dB. Only when displayinggraphs, the sound power is converted to sound power level. This to reduce complexity ofthe calculations and therefore calculation time.

During updating, there are some displays that are being updated. You are presented withthe time data,

The measured pressure,

And the calculated volume velocity,

Note that the images here are from running in Demo mode, and might differ alot from anactual tuning.


Tuning Errors

In case of divergence, the voltage could oscillate and produce high levels. This can occurwhen using wrong calibration data.

Before any loudspeaker is tuned, the Error panel displays,

If the analysis of the feedback signal pressure would get a mean level below 40 dB, thefunction indicates signal error and aborts the adjustment. This also sets the voltage signalto 0 V, to prevent unwanted amplitudes. A signal error could for instance mean that nomicrophone is connected to it’s respective input channel on the backend, malfunction inthe ICP amplifier or error in calibration data.

If the level would become 0, due to a tuning error, the function will indicate Total Error.

G.9. DATA PREPARATION 37

G.9 Data Preparation

After you have set up all sounds, assigned the loudspeakers and adjusted the levels, it’stime to prepare the sound data. This is done in the Data Preparation panel. Beforeany data is generated, the Preview Data, Preview Trigger and Show DAPL buttons aredisabled.

Here you specify the length of the playback. When playing longer than 1 second, theGUI will generate raw data files on the harddrive, used for streaming the data to theData Acquisition Processors. You also specify the Bandwidth used for playback. The bestresults is when using the maximum value, but when playing very long files, you mightneed to reduce the bandwidth due to memory resources.

When you have decided on a playback time and bandwith, press the Generate Data button,or use the menubar by going to Data Preparation → Generate Data.

After you have generated the data, you can preview the data by clicking on the Preview


Data button. This is only enabled when using a playback time below 10 seconds due toplotting of very large vectors. In the preview data window, you can also compare theloudspeakers to each other, and check the phase etc.

You can also preview the generated DAPL 2000 configuration files. This is mainly used fortroubleshooting the ENS.

Logfiles can be saved by pressing the Save Preparation Log, or enabling the Autolog feature,saving the log window upon every data generation process.

G.9.1 Generation Errors

When loading soundfiles, the sound file might need to be resampled to meet the require-ments of the DAP 5016a. If the input sound file has a sampling frequency which causesthe resampling filter to have a large complexity, you might need to change the internalresampling parameters in the Internal ENS Settings.

If you are creating very long playback files, there will be very high memory consumption.If you would get a memory error, you can lower the Bandwidth. This could howeverproduce some transient artifacts, but filtering will try to reduce the effect. There is a low

G.9. DATA PREPARATION 39

pass filter, of 6:th order, which will try to smoothen all transients above the bandwidth.If the Output Update Period is very large, this can however not be avoided even with thelow pass filter, as the artifacts occur in the hardware D/A1 conversion. The table below(table G.4), describes the different update periods and how they affect bandwidth, qualityand its artifacts.

Table G.4: Known Output Update Period quality and artifact creation.

Update Period Bandwidth Quality Artifacts1.20 8681 Best None1.25 8334 Very Good Nearly none1.50 6945 Ok Ok2.00 5209 Poor Some transients2.50 4167 Very Poor Noticable transients10.00 1042 Extremely Bad Machinegun

1Digital-to-Analog


G.10 Running the ENS

On the main ENS GUI window, there is the Run panel. From here you start and stop theENS, and you can also see various error messages and overviews in the logging window.Note that before you have prepared any data, you cannot start the ENS.

G.10.1 Setup Overview

To see an overview of the current ENS setup, click in the Setup Overview button. This willdisplay Simulation Time, Number of Repetitions, Sampling Frequency, Bit Resolution,Number of Sound Groups, Number of References and a list of all used and non-usedspeakers with their corresponding sound group.

There are more fields shown than described here.

G.10. RUNNING THE ENS 41

G.10.2 Starting the Engine

After you have generated the data, the Start Engine button is enabled.

When you are happy with the setup, you start the engine by clicking on the Start Enginebutton. This is a toggle button, so if you want to stop the engine you can click it while itis running.

If you have non tuned loudspeakers, or level warnings, you will be presented with a popupwindow. If you want to run the ENS anyway, discard the warnings.

First, the ENS is initialising, creating connections to the Data Acquisition Processors,flushing the pipes etc.

When the startup is completed, and no errors occured, the engine will start,

When the playback is completed, you will be presented with any errors, or a successmessage,


G.11. DEFAULT SETTINGS OF THE ENS 43

G.11 Default Settings of the ENS

G.11.1 Global Settings for Version 2.1

Setting Value

Simulation Time 1 second

Number of Runs 1 run

Sampling Frequency (1.20 · 10−6)−1/(2 · 24) HzNumber of Bits 16 bits (signed)

Integer Maximum ± 32767

Auto Save Preparation Log On

Auto Save Run Log On

Number of Groups 1 Group

Selected Group Group 1

Number of Sides 6 Sides

Selected Side 1 (Top)

Number of Loudspeakers 29 Speakers

Project Name Untitled Project

DAPID Loudspeakers 1 – 24 DAP0

DAPID Loudspeakers 25 – 29 DAP1

Active Loudspeakers to Input 0 & 1 Generated with Data Preparation

G.11.2 Default Sides Settings

Side Name Speakers

Side 1 Top 1 – 6

Side 2 Bottom 23, 24Side 3 Front 25, 26Side 4 Back 27 – 29

Side 5 Left 13 – 18, 21, 22Side 6 Right 7 –12, 19, 20


G.11.3 Default Sound Group Settings

Setting Value/Option Option Value

Name Group 1

Active Yes

Selected Sound Sinusoid

Sinusoid Frequency 1000 Hz

Phase 0 degrees

Noise Type White (Gaussian)

Sound File Path None

Data Empty

Sine Sweep Start Frequency 20 Hz

Stop Frequency 2000 Hz

Filter Type Off

Break Frequency Low 1000 Hz

Break Frequency High 2000 Hz

Order 4

Sound Data Empty Vector

Speakers 1 -- 29

G.11.4 Default Loudspeaker Assignments

Setting Value

Sound Group 1 (Group 1)

Reference No, it is not a reference

To Reference None

Active Yes

Relative Phase 0 degrees

Name Speaker #

Output Level 80 dB Sound Power Level

G.11. DEFAULT SETTINGS OF THE ENS 45

G.11.5 Default Calibration Settings

Speaker Volume Sensitivity Serial No

m3 mV/Pa

01 0.007 4.73 9073202 0.007 5.19 10650403 0.007 4.24 9464204 0.007 5.29 9464305 0.007 5.34 9464406 0.007 4.98 9464807 0.007 5.70 9464908 0.007 5.10 9465109 0.007 3.86 9465310 0.007 5.04 9465711 0.007 6.24 9465812 0.007 4.89 9466513 0.007 5.62 9466714 0.007 5.47 9466815 0.007 5.16 9467216 0.007 5.52 9467617 0.007 5.02 10648118 0.007 4.94 10648219 0.007 4.91 10648420 0.007 5.23 10648621 0.007 5.48 10648722 0.007 6.49 10648823 0.007 4.25 10649024 0.007 5.80 10649125 0.007 4.27 10649326 0.007 4.96 10649427 0.007 5.01 10946528 0.007 4.54 10649629 0.007 5.51 106497

G.11.6 Default Trigger Settings

Input Trigger Output Trigger

Setting Value Setting Value

Enabled Off Enabled Off

Level 10 V Level 10 V

Type Start Type Start

Function Pre/Post

Delay 1 second


G.12 Calibration Files Specification

G.12.1 Global File

The Import function reads a file which contains the folder architechure for the exporteddata. This could for example contain the file structure specification seen in the codesegment below. The function will start importing the files specified after START commanduntil End Of File (EOF) is reached. Note that any comments in this file can be writtenbefore the START command.

The files specified in this structural file should be in ascending order. It does however notmatter which type of data (REFERENCE or RESPONSE) comes first. Between START andEOF, all lines should should have the following specification,

[Element Number] [REFERENCE/RESPONSE] [Path]

01 REFERENCE Element01/Run01/Reference.mat

01 RESPONSE Element01/Run01/Response.mat





etc...

G.12.2 Exported Structures

The measurements to be exported are autopower spectrum of the reference microphone(inside the loudspeaker cavity) and all 20 microphones positioned on the ISO sphere (forcalculation of sound power level), for all loudspeaker elements. The settings for the mea-surements can be found in table G.5. The software expects 1 V to be the reference inputsignal, so if a different output amplitude is used, the measurements has to be modifiedmanually before importing them to the ENS.

Table G.5: Calibration Measurement SettingsI/O Setting Value

Overall Bandwidth 8192 HzResolution 1 HzSpectral Lines 8192Measurement Time 120 s (total)Number of Means 120Measurement Time 1 s (mean)Means LinearWindow Hanning

Output Type White NoiseAmplitude 1 V

Input Type AutoPowerSpectrumAmplitude Scaling RMSSpectrum Unit PaSpectrum Format LinearWeighting None

G.12. CALIBRATION FILES SPECIFICATION 47

The measurements should be exported to Matlab from Lms, by using the naming andfile structure given in table G.6, because the software is written to use the specific exportstructures used by Lms. These structures are defined as figure G.12.

Table G.6: Lms Naming and Exporting Filenames for Calibration

Entity Lms Name Exported Filename Contains Structure

Reference Reference Reference.mat AutoPower Reference S

Response (Sphere 1) POWER1 Response.mat AutoPowerSpectrumResponse (Sphere 2) POWER2 Response.mat AutoPowerSpectrumResponse (Sphere 3) POWER3 Response.mat AutoPowerSpectrum

. . . . . . Response.mat AutoPowerSpectrumResponse (Sphere 18) POWER18 Response.mat AutoPowerSpectrumResponse (Sphere 19) POWER19 Response.mat AutoPowerSpectrumResponse (Sphere 20) POWER20 Response.mat AutoPowerSpectrum

AutoPower Reference S

.x values

.start value 0

.increment 1

.number of values 8193

.y values .values [ 1x8193 ] (double)

AutoPowerSpectrum

.x values

.start value 0

.increment 1

.number of values 8193

.y values .values [ 8193x20 ] (double)

Figure G.12: The definition of the structures generated by Lms ”Export to Matlab“function. All unused fields are excluded from the structure for clarity reasons.

Documents

2010:150 CIV MASTER'S THESIS Engine Noise Simulatorltu.diva-portal.org/smash/get/diva2:1031701/FULLTEXT01.pdf · 2016. 10. 4. · 2010:150 CIV MASTER'S THESIS Engine Noise Simulator