Acoustics

Universit
at des Saarlandes
Spoken Language Systems
Development of Test and Measurement

Software for Electroacoustic Parameters
on Hearing Aids
Submitted to Saarland University

Faculty of Natural Sciences and Technology I
Department of Computer Science
in partial fulfillment
of the requirements for the degree of
Master of Science in Computer and Communications Technology
by
Mohamed Imran Noor Mohamed

Supervisor
Prof. Dr. Dietrich Klakow

Advisor
Dipl.-Ing. Oleg Fallmann

Dipl.-Ing. Uwe Nauerz
Reviewers
Prof. Dr. Dietrich Klakow

Prof. Dr.-Ing. Chihao Xu
January 2015
Eidesstattliche Erkl
arung
Ich erkl
are hiermit an Eides Statt, dass ich die vorliegende Arbeit selbststandig verfasst
und keine anderen als die angegebenen Quellen und Hilfsmittel verwendet habe. Ich
erklare hiermit an Eides Statt, dass die vorliegende Arbeit mit der elektronischen Version u
bereinstimmt.
Statement in Lieu of an Oath

I hereby confirm that I have written this thesis on my own and that I have not used any
other media or materials than the ones referred to in this thesis. I hereby confirm the
congruence of the contents of the printed data and the electronic version of the thesis.
........................................................
.................................................
(Ort und Datum / Place and Date)
(Unterschrift / Signature)
Einverst
andniserkl
arung
Ich bin damit einverstanden, dass meine (bestandene) Arbeit in gedruckter und elektronischer Form in die Bibliothek der Informatik aufgenommen und damit veroffentlicht
wird. Ich erkl
are dass mit einer derartigen Veroffentlichung keine Rechte Dritter verletzt
werden.
Declaration of Consent
I consent to make my thesis (with a passing grade) accessible to the public by having it
added to the library of the Computer Science Department in electronic and in printed
form. I declare that a publication of this kind does not infringe any rights of third
parties.
........................................................
.................................................
(Ort und Datum / Place and Date)
(Unterschrift / Signature)
ii
For their unconditional love and support, I owe this thesis to my family and almighty.
- Imran
iii
Acknowledgements
I would like to express my sincere gratitude to my supervisor Prof. Dr. Dietrich Klakow
for a wonderful learning opportunity he provided to me. I feel deeply indebted to the
way he patiently let me learn from my own mistakes. With his excellent supervision, I
have learned and improved a lot about the process of doing scientific research. Because
of his invaluable guidance, I have been able to groom myself as a better contributor
towards research. I consider myself very fortunate to have worked with him.
I would like to thank my mentor Oleg Fallmann for his invaluable guidance. I feel
deeply indebted for his strong advises, encouragements and instructions. He has not
only assisted me in completing this thesis, but also helped me to broaden my attitude
towards research, and in developing my personality.
I am highly grateful to Prof. Dr.-Ing. Chihao Xu for reviewing my thesis. I
would also like to thank advisor Uwe Nauerz for his technical guidance throughout the
thesis. I would like to take this opportunity to also thank all the members of Cetecom
organization for their support.
I am also very grateful to my friends and classmates for their company and moral
support. I learned a lot from them in last two years which is a very important factor for
successful completion of my thesis. Finally, I thank my almighty for all the good deeds.
iv
Abstract
For an effective test systems, having a reliable and efficient automated test and measurement solutions is of critical important. Especially for evaluation of hearing aids
having an accurate and time efficient calibration using an equalization algorithm is the
paramount objective to satisfy stringent requirements posed by IEC 60118 standards.
In the context of this Masters thesis we aimed to develop an automated test and
measurement software to evaluate electro-acoustical parameters on hearing aids by implementing a novel equalization algorithm to compensate unfavorable room effects on
the grounds of characterization of room acoustics.
The novelty of our developed equalization algorithm is the algorithm equalizes bad
room acoustics with an accuracy of +/ 0.5 dB and time taken for equalization is below
1 hour. Suitable experiments on room characterization finds the optimized positions
to place microphone and loudspeaker inside the room which significantly reduces the
reflections by adding acoustic absorbers.
The implemented novel equalization algorithm outperforms the default proprietary
solutions in terms of accuracy and time efficiency. We had developed a solution which
produces a reliable and accurate results for all sound pressure levels (90. 80. 70, 60
dB) except for less than 1% of all data points in 50 dB which is still considered acceptable. Hence, the developed solutions elevates the capability of next generation test and
measurement systems by satisfying performance metrics set by IEC standards.
Contents
Acknowledgements
iv
Abstract
vi
List of Figures
List of Tables
xii
Abbreviations
xiii
1 Introduction
1.1 Background . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
1.2 Motivation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
1.3 Outline . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
1
1
2
3
2 Problem Formulation
2.1 Need for Calibration . . . . . . . .
2.2 Equalization by R&S K-7 Software
2.3 Equalization by Cetecom software
2.4 Research Questions . . . . . . . . .
2.5 Plan of Attack . . . . . . . . . . .
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
4
4
6
7
8
9
3 Acoustics: Overview
3.1 Fundamentals of Sound . . .
3.2 Propagation of Sound . . . .
3.3 Sound Level and the Decibel
3.4 Acoustic Components . . . .
3.5 Room Acoustics . . . . . . . .
3.6 Absorption on Surfaces . . . .
3.7 Early Reflections . . . . . . .
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
10
10
11
12
13
13
14
15
.
.
.
.
16
16
16
18
19
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
4 Methodology
4.1 Software Design and Architecture .
4.1.1 SCPI: Overview . . . . . .
4.1.2 VISA Library . . . . . . . .
4.1.3 Event Driven Programming
.
.
.
.
viii
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
Contents
4.1.4
ix
MVC: Design Pattern . . . . . . . . . . . . . . . . . . . . . . . . . 20
5 Characterization of Room Acoustics

5.1 Room Acoustic Analysis . . . . . . . . . . . .
5.2 Reverberation Time . . . . . . . . . . . . . .
5.2.1 Theory . . . . . . . . . . . . . . . . .
5.2.2 Practical Measurements . . . . . . . .
5.2.3 Results . . . . . . . . . . . . . . . . .
5.3 Room Impulse Response . . . . . . . . . . . .
5.3.1 Theory . . . . . . . . . . . . . . . . .
5.3.2 Practical Measurements . . . . . . . .
5.3.3 Results . . . . . . . . . . . . . . . . .
5.4 Room Frequency Response Analysis . . . . .
5.5 Room Mode Calculation . . . . . . . . . . . .
5.6 Room Noise Analysis . . . . . . . . . . . . . .
5.6.1 Theory . . . . . . . . . . . . . . . . .
5.6.2 Results . . . . . . . . . . . . . . . . .
5.7 Evaluation of Room Characteristics . . . . . .
5.7.1 Reverberations and Impulse Response
5.7.2 Loudspeaker Evaluation . . . . . . . .
5.8 Proposal for Improvements . . . . . . . . . .
5.8.1 Adding Absorbers . . . . . . . . . . .
5.9 Summary . . . . . . . . . . . . . . . . . . . .
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
6 Equalization
6.1 Equalization Design and Working . . . . . . . . .
6.1.1 Programming Decision . . . . . . . . . . .
6.1.2 Theory and Block Diagram . . . . . . . .
6.2 Setup and Results . . . . . . . . . . . . . . . . .
6.3 Verification . . . . . . . . . . . . . . . . . . . . .
6.3.1 Verification at 50 dB SPL . . . . . . . . .
6.3.2 Statistical Accuracy at 50 dB SPL . . . .
6.4 Algorithm Runtime . . . . . . . . . . . . . . . . .
6.5 System Testing . . . . . . . . . . . . . . . . . . .
6.5.1 Comparison of Field Frequency Response
6.6 Summary . . . . . . . . . . . . . . . . . . . . . .
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
23
23
24
24
25
25
26
26
28
29
30
31
33
33
34
35
35
36
38
38
40
.
.
.
.
.
.
.
.
.
.
.
42
42
42
43
46
49
49
51
52
52
53
54
7 Conclusion and Future Work
55
Bibliography
57
List of Figures
2.1
2.2
2.3
Sound source calibration by R & S; K-7 software . . . . . . . . . . . . . .

Sound source calibration by Cetecom(Minimum Linearization) . . . . . .
Sound source calibration by Cetecom(Maximum Linearization) . . . . . .
3.1
3.2
Condensation and rarefaction effect of sound . . . . . . . . . . . . . . . . 11

Sound travel inside a room . . . . . . . . . . . . . . . . . . . . . . . . . . 14
4.1
4.2
4.3
4.4
4.5
Example SCPI command explaining device operation . . . .

VISA hierarchy with different communication bus protocols
Event driven programming with several processes . . . . . .
Model View Control . . . . . . . . . . . . . . . . . . . . . .
MVC- UML based design of modules . . . . . . . . . . . . .
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
17
18
19
21
21
5.1
5.2
5.3
5.4
5.5
5.6
5.7
5.8
5.9
5.10
5.11
5.12
5.13
5.14
5.15
5.16
5.17
Flow chart explaining RT30 measurement . . . . . . . . .

RT30 plot inside acoustic chamber . . . . . . . . . . . . .
Ideal RIR plot . . . . . . . . . . . . . . . . . . . . . . . .
Flow chart explaining RIR measurement . . . . . . . . . .
RIR inside the acoustic chamber . . . . . . . . . . . . . .
Room dimension for frequency response measurement . .
Frequency response measurement inside acoustic chamber
Room modes inside acoustic chamber . . . . . . . . . . . .
Noise explanation with system theory . . . . . . . . . . .
Noise inside the acoustic chamber . . . . . . . . . . . . . .
RT-30 plot inside the dead room . . . . . . . . . . . . . .
RIR plot inside the dead room . . . . . . . . . . . . . . .
Frequency response plot inside dead room . . . . . . . . .
Standard Deviation plot for KS Digital loudspeakers . . .
RIR: room without absorbers . . . . . . . . . . . . . . . .
RIR: Room improvement with absorbers . . . . . . . . . .
RT-30: Room improvement with absorbers . . . . . . . .
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
26
27
27
28
29
30
31
33
33
34
35
36
37
38
39
39
40
6.1
6.2
6.3
6.4
6.5
6.6
6.7
6.8
Equalization algorithm design: adaptation from

Equalization module flow chart . . . . . . . . .
Best case setup inside acoustic chamber . . . .
Equalization results for 90 and 80 dB SPL . . .
Equalization results for 70 and 60 dB SPL . . .
Equalization results for 50 dB SPL . . . . . . .
Verification at 50 dB SPL . . . . . . . . . . . .
Statistical Deviation with +/ 0.5 dB . . . . .
M.Sc seminar
. . . . . . . .
. . . . . . . .
. . . . . . . .
. . . . . . . .
. . . . . . . .
. . . . . . . .
. . . . . . . .
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
44
47
48
49
50
50
51
52
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
6
7
8
List of Figures
xi
6.9 Algorithm run time as size of data increases . . . . . . . . . . . . . . . . . 53

6.10 Field of frequency response: using R &S system . . . . . . . . . . . . . . . 53
6.11 Field of frequency response: using novel equalization algorithm . . . . . . 54
List of Tables
5.1
5.2
Room Dimensions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 25
Room Mode Calculation for Cabin . . . . . . . . . . . . . . . . . . . . . . 32
xii
Abbreviations
ISO
International Organization for Standardization
IEC
International Electrotechnical Commission
SPL
Sound Pressure Level
B&K
Bruel and Kaer
R&S
Rhode and Schwarz
SCPI
Standard Commands Programmable Instruments
IVI
Interchangeable Virtual Instruments
VISA
Virtual Instruments Software Architecture
COM
Communication
MVC
Model View Controller
RT
Reverberation Time
DSP
Digital Signal Processing
RIR
Room Impulse Response
xiii
Chapter 1
Introduction
In the next-generation test and measurement systems, automated test equipment and
solutions are critical components that determines the effectiveness and efficiency of the
overall test system. The automated test solutions are envisioned to offer a variety of advantages over the current systems in terms of flexibility, cost effectiveness and reliability.
However, when we consider evaluation of hearing aids performance using automated test
solutions the electro-acoustical measures form the basis [1]. The realization of an effective electro-acoustical test and measurement system for hearing aids requires a rigorous
interplay between acoustic knowledge and the understanding of the existing standards,
as effects of electro-acoustic characteristics has an elusive link to behavioral aspects for
the hearing aid users [2].
1.1
Background
As one of the prerequisite to achieve, a fast, reliable and effective electro-acoustical measurement is calibration of the acoustic components such as microphone and loudspeakers. Although, the techniques to calibrate microphones is trivial, yet current systems
and methods for calibrating loudspeakers with a connection to a microphone located at
a listening area in a room are often limited to manual calibrations. The development
of a self-calibrating solutions for loudspeaker includes specific features to adjust speaker
characteristics based on the effects generated by operating the loudspeaker in the room.
Under this scenario, equalization for room acoustics plays a vital role to produce desired
1
Chapter 1. Introduction
sound field inside the room [3]. The term Equalization has broad meaning, in specific,
we consider to adjust or linearize the frequency components produced inside the acoustic
room.
As an example, the microphone picks up a test signal generated by the loudspeaker
and the system uses the test signal to determine the loudspeaker frequency response.
Thus, the frequency response is analyzed and characterized. Based on the analysis, the
system generates parameters to compensate unfavorable effects inside the room with the
help of an efficient equalization algorithm. This constitutes the key to automate tasks
which, in certain cases, requires accuracy and time-efficiency.
The International Electro-technical Commission(IEC) has published several standards for various applications within hearing aids. Of particular importance is IEC
60118, which constitutes the standard for the measurement procedure for hearing aids
[4]. Hence, to reproduce precise electro-acoustical characteristics on hearing aids the
accuracy in the equalization as technique for calibration is to be +/ 0.5 dB(decibel).
For deviations higher or lesser than 0.5 dB seems to alter the frequency response on
the hearing aids. Similarly, having an effective measurement time to evaluate hearing
aids also needs to be considered for practical feasibility. Although, the description of
accuracy and time efficiency exactly does not correlate to IEC standards. However,
to achieve standard compliance the automated test solution need the aforementioned
requirements for effective evaluation of the device under the test.
1.2
Motivation
Various equalization techniques have been proposed for implementing the room response
compensation such as DSP(Digital Signal Processing) based techniques by Karjalainen
et al. [5] and other techniques involving modal equalization [6]. Yet, in the context of the
thesis, the development of an equalization algorithm considers the stringent requirements
in accuracy of +/ 0.5 dB and measurement time efficiency within 1 hour as a basis to
satisfy IEC standards. Similarly, our requirement is to develop an equalization based
on a non post-processing algorithm as extra signal processing will influence hearing aids
characteristics. This requirements brings up novelty and challenges in the thesis to
automate test and measurement systems especially for hearing aids.
Chapter 1. Introduction
The present thesis project is conducted at the Cetecom ICT Services GmbH, Saarbrucken with a specific innovation objective to investigate and implement efficient room
response equalization, making the automation of electro-acoustical measurements reliable using existing tools.
1.3
Outline
Chapter 2 explains the problem addressed in the Masters thesis and the research
questions that are posed.
Chapter 3 describes the basic theory of acoustics and room acoustics.
Chapter 4 explains the methodology to design software.
Chapter 5 explains the characterization of room acoustics.
Chapter 6 explains the design and implementation of the equalization algorithm
with system testing and verification.
Chapter 7 explains the the summary of our work with possible opportunities of
the future work are discussed.
Chapter 2
Problem Formulation
Development of an efficient test and measurement solution for hearing aids measurements
are critical. The entire application development involves synchronization and triggering
of measurement routines to process measurement data and error management. Taking
all these parameters into account, the development of an instrument control software
presents several challenges.
This chapter describes the problem that is addressed in the present Masters thesis
and the research questions that are posed.
2.1
Need for Calibration
Acoustic measurements often involves measurement of sound pressure level or the generation of a sound field, or both. Typical examples are noise measurements, loudspeaker
measurements, microphone measurements and measurements on systems like hearing
aids and mobile phones. A sound level meter, for example, is supposed to directly measure the sound pressure level in dB SPL (dB referred to 20 P a sound pressure). If an
audio analyzers(such as Rhode & Schwarz UPV) is used for this purpose, the sensitivity
of the measuring microphone Sm
Sm = Vm /p
(2.1)
Chapter 2. Problem Formulation
has to be determined, wherein Vm is the RMS value of the microphone output voltage
and p is the RMS value of the acoustic pressure on the membrane. From the equation 2.1,
the sound pressure is obtained by dividing the measured voltage by this sensitivity value.
The measurement by which the value of the microphone sensitivity is obtained is known
as microphone calibration. The sound pressure level for the microphone calibration is
generated with an acoustic calibrator (such as B & K) which generates a sine wave signal
with defined frequency (commonly 1 kHz) and defined sound pressure (usually 1 Pa or
10 Pa). As measurement microphones are small and have a well-defined mechanical
structure, the sensitivity is frequency-independent within a 4k Hz till 20k Hz frequency
range. Therefore, calibration at a single frequency is sufficient.
However, certain measurements requires sound generation using loudspeakers in a
defined sound pressure at a certain point. Thus, loudspeakers sensitivity is a measure
of Sound Pressure Level (SPL) at a specified distance for a specified input signal in RMS
voltage, typically at one or more specified frequencies (often 300, 400, 500, 600 Hz or
the average of these).
As a loudspeaker has a far more complex mechanical structure than a measuring
microphone, and radiation effects additionally influence the loudspeaker sensitivity depending on the frequency and on the location of the measurement point, the sensitivity
of a loudspeaker usually is frequency dependent. For this reason the calibration of a
loudspeaker consists of two steps:
Measurement of the absolute sensitivity at one frequency

Measurement of the frequency response relative to this frequency.
In order to generate a defined sound pressure level at the measurement point, the
generator output sound pressure has to be set to the desired sound pressure divided by
the loudspeaker sensitivity, and to be corrected by an efficient inverse frequency response
as known as equalization.
This process of producing required sound pressure level from generator or loudspeaker
involving equalization is known as loudspeaker calibration. The Equalization plays a
crucial role in finding desired sound pressure level (dB SPL) for an applied voltage(dBV)
at different frequencies.
Since calibration is the first procedure that needs to be conducted before any measurements. Having accurate and efficient calibration is important especially for conformance
testing procedure as explained by hearing aids standards such as IEC 60118.
2.2
Equalization by R&S K-7 Software
To manage and record calibration values for acoustic devices such as sound source, microphone and ear simulator automatically by means of suitable routines R & S developed
K-7 software. However, while performing calibration of the sound source the K-7 software misses the accuracy to equalize the sound pressure well within the range of +/- 0.5
dB SPL, which is a stringent requirement from the hearing aid standards IEC 60118.
Figure 2.1: Sound source calibration by R & S; K-7 software
As an illustration, we did equalization for the sound source (loudspeaker) on sound

pressure levels 50, 60, 70, 80, 90 dB SPL using K-7. As shown in Figure.2.1 after 3000
Hz till 9000 Hz the equalization of the sound pressure levels (in dB SPL) is not accurate
and it deviates up to 1 dB SPL in all sound pressure levels (50 dB SPL till 90 dB SPL).
Therefore, the results shows that equalized values does not satisfy requirements set
by standards. With this accuracy, further measurements involving hearing aids would
produce less accuracy and the whole compliance cannot be certified for the device under
test as the test solution is not accurate enough adhering the standard requirements.
Hence, the entire equalization solution becomes a bottle neck for hearing aids testing
within the acoustic lab. Therefore, we need a novel research solution to solve the equalization problem within the standards maintaining both accuracy and time efficiency.
2.3
Equalization by Cetecom software
To solve the equalization problem, Cetecom developed an algorithm to linearize the

sound pressure values. Initially, they developed a software with accuracy greater than
+/- 0.5 dB SPL and named as Minimum Linearization algorithm which equalizes the
sound pressure level in a well defined time (less than 30 minutes). Yet, the algorithm is
less accurate as shown in Figure.2.2
Figure 2.2: Sound source calibration by Cetecom(Minimum Linearization)
However, to increase the accuracy of the algorithm they developed Maximum Linearization algorithm by increasing the iterative procedure used in the software for much
longer cycles. Hence, the new developed solution is accurate but not time efficient
(greater than 1 hour) as shown in the Figure.2.3
Being time efficient is not a requirement from the standard for performance evaluation. Yet, for typical hearing aid measurements we need a time budget to start and stop
test procedure well within a time frame. As without time efficient solution the software
becomes impractical.
Under this scenario, we need an equalization algorithm which linearizes the sound
pressure levels within an certain accuracy (+/- 0.5 dB) and be time efficient (typically
less than 1 hour)
Figure 2.3: Sound source calibration by Cetecom(Maximum Linearization)
2.4
Research Questions
In the context of the thesis, several research questions are posed concerning the development of accurate and time efficient equalization algorithm to correct bad room acoustics
for further hearing aid measurements.
What are the basic characteristics and quality assessment can be done for the
acoustic chamber to develop an efficient equalization algorithm?
What possible improvements can be achieved feasibly based on the detailed analysis for the acoustic chamber?
Which is the proposed equalization solution for overcoming the challenges in terms
of accuracy and time efficiency?
What are the potential limitation and bottlenecks for the developed solution in
the context of hearing aid measurements that occur while satisfying the stringent
requirement of IEC hearing aid standards?
The theoretical research and implementation of solution was conducted in the context
of the present Masters thesis. It aims to answer the above mentioned questions in a
detailed manner. The thesis involves development of efficient equalization solution for
hearing aid measurement by detailed room acoustic analysis.
2.5
Plan of Attack
To solve and realize our problem formulation in a systematic way. We structure our
approach into four particular stages namely:
Software Design and Architecture: The set of operations defined by the measurement instruments should be compatible with our software development. Hence,
a brief understanding is required to map our software into the system. Here we
understand the resources and libraries used for mapping in Chapter 4
Characterization of our acoustic chamber: To provide scientific solution to our
problem, we need to understand the properties of our room with respect to acoustics. Thus, behavioral measurements will lead us to develop modules efficiently
and accurately corresponding to our scenario in Chapter 5
Equalization for room non-linearities: An equalization algorithm is designed and
implemented in a way that the performance of the algorithm is guaranteed for its
efficiency and accuracy. Therefore, statistical accuracy and algorithm run time
is taken into consideration with high priority to verify the algorithm for its best
effort performance in Chapter 6
System Testing and Verification: The implemented equalization algorithm is tested
for its system functionality and evaluated with the default scenario using R&S K-7
software in Chapter 6
Chapter 3
Acoustics: Overview
In the present chapter, the basic theory of acoustics and in particular room acoustics
terminology are presented. This chapter covers the essential theory acquired during the
phase of thesis based on which characterization of the room for efficient equalization is
developed.
3.1
Fundamentals of Sound
Sound is defined as vibration transferred through a medium. For example, if an instrument such as loudspeaker sounds, the vibration from the instrument is transmitted
into another medium which is the air. From a physical point of view, the instruments
vibrations are pushed as a pattern of condensation and rarefactions into the air. This
pattern is further transmitted to the surrounding air and hence a longitudinal wave is
created by a vibrating object. Thus, the wave carries the pattern of vibrations through
the air at approximately 343 meters per second(m/s) at 20 C.
To illustrate, how sound waves are produced and why they are longitudinal, consider
the vibrating diaphragm of a loudspeaker. When the diaphragm moves outward, it
compresses the air directly in front of it, as shown in Figure.3.1. This compression
causes the air pressure to rise slightly. The region of increased pressure is called a
condensation, and it travels away from the speaker at the speed of sound.
After producing a condensation, the diaphragm reverses its motion and moves inward,
as in Figure.3.1 (part b). The inward motion produces a region known as a rarefaction,
10
Chapter 3. Acoustics: Overview
11
Figure 3.1: Condensation and rarefaction effect of sound
where the air pressure is slightly less than normal. Following immediately behind the
condensation, the rarefaction also travels away from the speaker at the speed of sound.
3.2
Propagation of Sound
From basics of sound, we understood that sound is a vibration that propagates through
a medium. Hence, sound can be created or transmitted only in a medium such a gas,
liquid or solid. Moreover, the particles present in the medium must disturb the wave to
move from one place to another. Thus, sound cannot exist in vacuum.
To understand sound propagation it is essential to know physical properties of sound
such as frequency and wavelength. The frequency of a waveform is how many full cycles
of the waveform we hear in a second. Its value is one divided by the period, where the
period is the amount of time it takes for the waveform to complete a full cycle as defined
in cycles per second or Hertz (Hz). In general, wavelength is a measure of the distance
between repetitions of peaks in a waves cycle. It is defined as :
= V /f
(3.1)
Where is the wavelength and V is the phase speed of the wave and f is wave
frequency.
12
Like any other waves, sound waves undergo certain behaviors when it encounters the
end of the medium or obstacles. Possible behaviors include reflections off the obstacle,
diffraction around the obstacle, and transmission (accompanied by refraction) into the
obstacle or new medium. For instance, diffraction happens when the waves bends around
small obstacles.
3.3
Sound Level and the Decibel
As compared to other physical units and measurements in acoustics, the intensity of

sound is measured in decibel (dB). Understanding decibel and its related units will help
us to explore into the area of acoustics and sound. In principle, the sound intensity is
defined as the sound power per unit area. The intensity I is measured as Decibel(dB)
scale.
I(dB) = 10 log10 [I/I0 ]
(3.2)
Where I0 = 1012 W atts/m2 ,

By definition, decibel is the ratio of a given intensity to the threshold of hearing
intensity, where this threshold takes the value 0 dB. The logarithm involved in the
formula is just the power of ten of the sound intensity as a multiple of the hearing
intensity. For example, if I/I0 is 104 then equivalent sound intensity is 40 dB. Similar
to the sound power we can measure sound level as L in dBV by applying voltages as v
and v0 , where v0 is referenced to 1 V.
By rearranging,
L(dBV ) = 20 log10 (v/v0 )
(3.3)
V = V0 10L(dBV )/20
(3.4)
The above relations between the applied voltage (in V) and produced sound pressure
(in dBV) is important to derive right sound level.
3.4
13
Acoustic Components
A microphone is an acoustics-to-electric transducer which converts the sound pressure

into an electric signal. There are different varieties of microphone available but for our
practical measurements a precise pressure sensing condenser microphones are used. It
uses a constant electrical charge to convert the diaphragm displacement into an analog
electrical signal.
A loudspeaker is a device used to convert an electric signal into acoustic waves. Loudspeaker is one of the important variables which may alter the frequency response of the
chamber which needs to be linearized. In principle, ideal loudspeakers produce acoustic
waves that are a linear transformation of the electrical input signal. Thus, it could be
seen as a LTI (Linear time-invariant) system. However, it is also well known that loudspeakers produce non-linearity which can be modeled mathematically [7]. Even though
loudspeaker manufacturers takes the non-linearity measures into consideration. Yet, the
actual cause and symptoms for non-linearity in different loudspeakers is well discussed
by Klippel, Wolfgang [8]. Also, compensation for the non-linearity in loudspeaker is a
topic of its own research as investigated on M.Sc Seminar.
In thesis, while developing equalization algorithm the acoustic characteristics produced from the loudspeaker is considered extensively before calibration is done. Yet,
internal characteristics of loudspeaker is not considered. Hence, we model loudspeaker
and other acoustics components such as amplifiers as black box model during development.
3.5
Room Acoustics
Consider the propagation of sound in an enclosed room. The sound conducting medium
is bounded on all sides of wall, floor and ceiling. These room boundaries are not ideal
surfaces, they absorb a portion of the sound energy impinging on them, and reflect the
rest. The absorbed energy is either converted into heat, or transmitted to the outside
of walls. These numerous reflected components is responsible for what is known as the
acoustics of a room.
14
If an acoustic wave hits a hard flat surface, it is reflected back (in Figure.3.2) into
the room due to the large difference in the acoustic impedance between the air and the
material of the hard surface.
Figure 3.2: Sound travel inside a room
The phenomenon of having reflected sounds in an enclosure is known as Reverberation.

The time taken for the entire activity is known is Reverberation Time (in Section 5.2)
3.6
Absorption on Surfaces
Real surfaces absorb a portion of the energy of the incident sound wave. Absorption
usually depends on the frequency. The absorption coefficient is the ratio between the intensity Ii of the incident wave and the intensity difference between incident and reflected
wave Ir :
= (Ii Ir )/Ii
(3.5)
An open window in a room has more or less 100% absorption ( = 1 ) at least for
frequencies where the wavelength is small compared to the dimensions of the window.
Hence, choosing an appropriate absorbing material will reduce reflections happening
inside the room.
3.7
15
Early Reflections
A sound impulse starts from a sound source traveling at equal speed in all directions. In
different distance from the sound source at different directions, the impulse hits a hard
surface of a wall or other obstacle in the room and is reflected.
At a certain point of observation, the direct sound traveling the shortest distance from
the source to the point of observation arrives first. One after the other, reflections arrive
from different directions, delayed according to the additional distance the impulse had
to travel due to its detour. If the sound intensity at the point of observation is plotted
over time, a so called reflectogram(Room Impulse Response) is obtained. Based on the
Room Impulse Response, the exact positions of reflections happening inside the room
can be found, which is conducted as an experiment in Section 5.3.
Chapter 4
Methodology
In the following sections, we explain the software design and architecture used in our
solution. Although it is not the aim of this thesis to study and understand the technical
details of a real platform and instruments used. Yet, in order to implement a running
prototype, having a brief knowledge on architecture and tools used is a requirement.
4.1
Software Design and Architecture
In context of the thesis, to meet all the technical and operational requirements with
feasibility, while optimizing essential quality attributes such as performance we need to
define a structured solution. Hence, software design and architecture encompasses the
required set of significant structure to develop good software.
4.1.1
SCPI: Overview
For effective communication between the PC and instrument via IEEE 488 bus standard commands is required. Thus, the SCPI- Standard Commands for Programmable
Instruments was defined by IVI (Interchangeable Virtual Instruments) foundation [9]
to have a generic command to communicate between instruments. In principle, SCPI
defines the roles of instruments and controllers in a measurement system.
The following are advantages of using SCPI:
16
Chapter 4. Methodology
17
Usability - Easy to query the applications as it uses template style which makes it
generic.
Consistency - From the remote programming standpoint, it offers a consistent communication between instruments of the same class with good functional capability.
Maintainability - Easy to maintain and extend commands.
Let us understand how SCPI commands are used in our software with an example to
appreciate its advantages.
Example: Sub Routine : IntializeSystemSetttings
Figure 4.1: Example SCPI command explaining device operation
In the Figure 4.1, a subroutine uses SCPI commands to control the instruments
such as in line 2, the commands instructs the instrument to put the display ON while
measuring data and then it switches the device output to ON state. Finally, it asks for
the device to return Boolean values as result of operation completion of the previous
executed commands. Thus, the above commands instruct the instruments with less effort
for us to memorize specific commands. Furthermore, it brings advantages to formulate
commands based on the structure provided by the standards.
4.1.2
18
VISA Library
Virtual Instrument Software Architecture, commonly known as VISA, is a widely used

input-output(I/O) API (Application Program Interface) to communicate with instruments and the PC. Generally, for test system development we require a software architecture that allows easy interchangeability of instruments between different systems.
Thus, using open industry standard software architectures such as VISA, we are able to
create systems with interchangeable test instrumentation [10].
The VISA standard library includes specifications for communication between resources over device specific I/O interfaces such as GPIB (General Programmable Instrumental Bus) or IEEE 488 bus. For our application, we used VISA architecture with
.NET framework where the library interacts over the Microsoft communication (COM)
technology.
Figure 4.2: VISA hierarchy with different communication bus protocols
In the Figure.4.2, the VISA library stands as gateway to communicate between instruments with different bus protocols. In our scenario, we use GPIB over 488.2 specification
for communication. The diagram exemplifies VISA library offering as :
An abstraction to differentiate between buses,
Incorporates different bus protocols and classes with the good functional capability,
19
Finally, it provides a unified API to communicate between test equipment-regardless

of communication bus.
Hence, we employed the VISA- software architecture to communicate between our

instruments efficiently.
4.1.3
Event Driven Programming
To test any application there is a well defined sequence (events) or standard procedure by
which system testing is needs to be done continuously. With this test sequential model,
we need to develop software that is reusable, maintainable and extensible corresponding
to various sequence (events) while testing. These features will also enable the software
to be adaptable to plug out old implementations and plug in new implementations with
ease. So, there is need to develop a system corresponding to events as it elevates the
possibility to quickly test the research ideas for real implementation.
Figure 4.3: Event driven programming with several processes
Hence, to achieve the above mentioned paradigm we need Event-Driven programming.

It is a programming technique that corresponds to different events based on the action
intended. In an event-driven application, there is generally a main loop that listens for
events, and then triggers functions when one of those events is detected. It can be best
understood with the help of an example.
Let us look at the Figure.4.3. a user executes a single process indicated in Green
circle, the single process calls several other sub-process indicated as Red and Violet
20
circles and the whole event is repeated until the successful final event or when a dead
lock happens Orange circle.
In an event-driven architecture, information can be propagated in near-real-time
throughout as a highly distributed environment. It promotes low run time of an application which is a requirement for our application development in Section 2.4. Also,
this type of programming techniques helps us to handle tasks sequentially according to
the test plan.
4.1.4
MVC: Design Pattern
Based on the event driven programming paradigm explained in the above paragraphs,
Model-View-Controller (MVC) is the right choice for the software design pattern. As
this pattern separates the modeling of the domain, the presentation, and the actions
based on user input into three separate classes [11]:
Briefly, the MVC contains:
Model: The model manages the behavior and data of the application domain,
responds to requests for information about its state (usually from the view), and
responds to instructions to change state (usually from the controller).
View: The view manages the display of information.
Controller: The controller interprets the mouse and keyboard inputs from the user,
informing the model and/or the view to change as appropriate.
Figure.4.4 depicts the structural relationship between three objects. It is important

to note that both the view and the controller depends on the model. Hence, a ModelView-Controller is a fundamental design pattern for the separation of user interface logic
from requirement logic corresponding to events.
One of the key advantage of using the MVC design pattern is testability. Testing
components becomes difficult when they are highly interdependent, especially with user
interface components. These types of components often require a complex setup just to
test a simple function. Worse, when an error occurs, it is hard to isolate the problem to a
21
Figure 4.4: Model View Control
Figure 4.5: MVC- UML based design of modules
22
specific component. This is the reason why separation of concerns is such an important
architectural driver.
Thus, MVC separates the concern of storing, displaying, and updating data into three
components that can be tested individually. So, the use of MVC design pattern is quiet
applicable for our event driven development.
Based on the understanding of MVC-design pattern, we implemented each modules or
class using this model as it was feasible. As an outcome, the Figure.4.5 has each module
designed using MVC-pattern. Finally, all components or modules are aggregated into a
single unit.
Chapter 5
Characterization of Room
Acoustics
This chapter covers the characterization of the room with suitable experiments.
5.1
Room Acoustic Analysis
From our problem statement in Section 2.4, room characteristics will give more insight to
understand rooms behavior with respect to sound. Therefore, understanding essential
room qualities will help us to design effective equalization solution with scientific goals.
The following sections, answers the below questions in an elaborate way with experimental results:
Does the room have reflections?

Where are the reflections inside the room?
What is frequency response with respect to reflections in the chamber?
What is the noise level in the room without acoustic source?
23
Chapter 5. Characterization of Room Acoustics
5.2
24
Reverberation Time
Reverberation Time is defined as the time required (in seconds), for the average sound in
a room to drop by 60 decibels (dB) after switching off the sound source. Reverberation
time indicates the presence of reflections inside a room. A Room with higher Reflections
is considered live than room with less reflections Dead.
5.2.1
Theory
Let us consider, we excite a sound source inside a room of Volume V . After sound in the
room saturates, we switch off the sound source: The reverberation time (RT60) obtained
using Sabine formula 5.1 is:
RT 60 = 0.16
Reverberation Time = Constant *
V
A
(5.1)
Volume / Absorption Area of the room
where absorption area of the room is A = S, where is the average absorption

coefficient of the room.
As equation 5.1 explains reverberation depends directly on volume and indirectly
on the area of the room. However, the above equation holds true only for the live
room having only one absorption coefficient . However, a room having more than
one absorption coefficient Erying [12] formula holds true to find reverberation time in
milliseconds(ms):
RT 60 = 0.16
V
S ln(1 )
(5.2)
where,
=
1X
Si i
S
(5.3)
equation 5.3 calculates the absorption coefficient for different dimensions such as
normal walls, walls with windows and so on as indicated by i. Hence, for our room with
25
Table 5.1: Room Dimensions
S.no
1
2
3
4
Room Parameter
Length
Breadth
Height
Surface area
Dimension
2.84 m.
3.06 m.
1.95 m
40.49 m2
less reverberation, plugging the values of room dimensions as described in the Table 5.1
in equation 5.3, we get theoretical reverberation time as 25 milliseconds(ms).
To verify our theoretical calculation a verification is conducted; the following section
will describe the brief method to measure reverberation time.
5.2.2
Practical Measurements
Initially, test setup was placed using ISO-3382 standard [13]. Figure.5.1 explains the
procedure to capture RT30. Briefly, RT plots the decay of sound over the time. During
measurement, we observed RT30 by calculating decay between -5 dB to -35 dB and
interpolated them for RT60.
For signal generation, we used sine burst tone as the excitation signal [14]. Sine
burst tones is repeated sinusoidal tone with sharp attack and decay. For a room with
reflections, the sine burst tone remains detailed. It was also easy to capture sound decay
as compared to other sound signal like IIR (Integrated Impulse response) using balloon
burst [15]. Further, the burst tone had good Signal to Noise Ratio(SNR) which is crucial
for reverberation time measurement.
For verification, we measured RT30 for 25 trails over the entire frequency sweep. This
additional measurement gave statistical description to our experiment.
5.2.3
Results
As Figure.5.2 shows that reverberation time (RT30) of our acoustic cabin is 11.5 milliseconds(ms). Hence, our RT60 is 23 milliseconds(ms). Also, the figure reveals a prominent
fact that reverberation time is higher at lower frequencies (100-1000 Hz) as compared
to the higher frequencies. Yet, it is impossible to conclude analysis based on single
experiment.
26
Figure 5.1: Flow chart explaining RT30 measurement
To find maximum reflections and where exactly reflections happen inside the room,
we will discuss about the Room impulse response in the following sections.
5.3
5.3.1
Room Impulse Response

Theory
In signal analysis, the room impulse response is the output response shown by a system,
when excited with an impulse input inside a room. The input impulse is characterized by
a Dirac delta functions [16]. As we know the presence of reflections inside the chamber
using RT30 measurement. Adding Dirac delta impulse to our room will detail out the
reflections with respect to environment geometry. Further, reverberations are linear and
time-invariant. Hence, the system is characterized by the given impulse response.
Mathematically, the output of this system, y(t), can be described as a convolution
between the systems input signal, x(t), and the room impulse response, h(t).
27
Figure 5.2: RT30 plot inside acoustic chamber
y(t) = (x h)(t)
Output Response = Input Signal * System Response
Figure 5.3: Ideal RIR plot
(5.4)
28
Using the above equation 5.4, the impulse can be calculated. Ideally, the impulse response plot should like in the Figure.5.3 [17]. The below paragraph explains the experimental procedure to measure RIR.
5.3.2
Practical Measurements
Figure.5.4 explains the logical procedure where Dirac signal is defined as :
+1, x = 0
(x) =
0, x 6= 0
(5.5)
From the above equation 5.5, it is clear that signal is defined at x = 0 and zero
elsewhere. Using this definition a Dirac wave file(.wav) is created as arbitrary function
in the measurement instrument with one unique value and zero elsewhere.
Figure 5.4: Flow chart explaining RIR measurement
5.3.3
29
Results
Figure.5.5 illustrates that the output response is not direct input impulse rather a response with variations indicating the presence of the reverberations at certain positions.
For example, at 8 milliseconds(ms) there is a maximal peak followed by lesser peaks.
From this, we can find the distance as:
Speed of Sound = Distance traveled by a sound wave
/ Time taken for sound travel
Using the above definition, the location of reflections is found. As an example, the
maximal peak 8ms corresponds to glassed mirror at 2.3 meters. Similarly, for time 5.5
ms, the distance of reflections were found to be at 1.87 meters (roof) and other smaller
reflections came from wall behind microphone.
Based on this measurement, we found the reflections happening at certain positions.
Yet, we could not figure out the actual cause for this reflections and what is the possibility
for the distortions inside the chamber. Hence, in the following section we will explore
more about the frequency response and possible cause for acoustic distortions.
Figure 5.5: RIR inside the acoustic chamber
5.4
30
Room Frequency Response Analysis
Frequency response analysis is a method to compute output response from a system

with respect to a steady state oscillatory excitation. The steady state excitation is a
sinusoidal wave. In frequency analysis the excitation is explicitly defined in frequency
domain. The main objective to measure frequency response is to reproduce the input
signal without any distortions. One feasible way to determine frequency response is to
feed a swept sine wave into an acoustic system and measure the output response [18].
The measurement procedure is same as previous sections. The standard test setup
is observed as illustrated in the below Figure.5.6 and the test signal is sent and output
response is measured.
Figure 5.6: Room dimension for frequency response measurement
The Figure.5.7 illustrates the output frequency response from the system. It is clear
that the frequency response is not flat and has distortions. Few possibilities for this
distortions such as:
Due to Reflections in the chamber: Based on the exploratory measurements on

reflections inside the room, the results from RT30 and RIR confirm the possibility
for reverberations inside the chamber. This effects are produced from the walls in
the acoustic cabin which produces unfavorable effects on the frequency response.
31
Room modes at lower frequencies: Another possible cause is where the sound
waves reflected between the walls in the room and interfere with each other. This
phenomenon is called resonance or room modes, as discussed in Section 5.5. From
theory, every room has got room modes at certain frequencies, where a standing
wave appears [19].
Loudspeaker and audio amplifiers characteristics: The characteristics corresponding to the instruments used to produce sound inside the chamber, as discussed in
Section 5.6.
Figure 5.7: Frequency response measurement inside acoustic chamber
5.5
Room Mode Calculation
Room Modes is an important acoustical phenomenon, which can have a significant impact on the rooms characteristics. In general room modes are modal frequencies (so
called eigenfrequencies) of a room with any shape. Hence any room has got room modes
at certain frequencies, where a standing wave appears. Standing waves are caused by
a perfect constructive interferences of sound waves, which are traveling between two or
more room boundaries [20]. The standing waves appear for several modal frequencies of
a room and can be derived with the basic wave equation.
fres
c
=
2
r
n
x
32
n
y
n
z
(5.6)
Where: f = frequency of the mode in Hz, c = speed of sound 343 m/s, nx = numbers
of natural oscillations (room length) (1, 2, 3, ...), ny = numbers of natural oscillations
(room width) (1, 2, 3, ...), nz = numbers of natural oscillations (room height) (1, 2, 3,
...), L, B, H = length, width and height of the room in meters
A standing wave in the 1st order of the fundamental frequency = f 1 occurs, when
half the wavelength of the excitation frequency fits between the sonically hard boundary
surfaces.
We can calculate standing waves of higher order from the integer multiples of the
1st order mode as:
f2 = 2 f1 ; f3 = 2 f2 ; f4 = 2 f3
(5.7)
The complete table is generated for all room modes corresponding to our room using
equation 5.6 and 5.7 :
Table 5.2: Room Mode Calculation for Cabin
Length=2.84 m
60.24
121.27
181.91
242.54
303.18
Breadth=3.06 m
56.8
112.55
168.83
225.10
281.38
Height=1.95 m
88.31
176.62
264.93
353.24
441.55
Graphically Figure.5.8 illustrates the distribution of the standing waves sound inside
the room. A considerable number of isolated room modes is present for low frequencies,
which might have an enormous impact on room characteristics. Hence, this factor is
taken into consideration to avoid them effectively.
33
Figure 5.8: Room modes inside acoustic chamber
5.6
5.6.1
Room Noise Analysis

Theory
The sound which is not desired is termed as Noise. The main purpose to measure noise
inside the room is to determine its influence on the room response. To understand them
better, let us model the noise function using system theory.
Figure 5.9: Noise explanation with system theory
34
The model in Figure.5.9 illustrates the situation where the measured signal b(t) includes
not only the signal v(t), the output signal from the system in response to the a(t),
but also some additive uncorrelated noise n(t). In theory n(t) might also include some
components transmitted from the system, but stemming from sources other than s(t).
Using this model, we decided to plot the actual noise present inside the room.
5.6.2
Results
The noise plot is shown in the Figure.5.10.
Figure 5.10: Noise inside the acoustic chamber
To find the frequency spectrum in which the entire noise is distributed, a suitable
frequency sweep is made with no input to loudspeaker. The above diagram clearly
indicates the noise level is more than 10 dB at 200 Hz and goes up to 15 dB at 300400 Hz frequency. This proves the presence of distortions in our room especially in
low frequency region. Although, this analysis indicated the noise yet modeling noise to
implement in equalization algorithm is beyond the scope the thesis. Hence, noise effect
is used only for analysis than compensation.
5.7
35
Evaluation of Room Characteristics
Acoustics quality evaluation broadly is the area of audio engineering which includes
various facets of evaluation from the sound perception to compression algorithms. In
general, quality is the measure of distance between the character of an entity under
study and the character of a target associated with this entity.
Given the fundamental definition, we took the acoustic characteristics in a room
as the basic entity and compared with our target dead room present in the Saarland
University to measure quality. Although, the dimensions of the dead room is larger
compared to the acoustic cabin in Cetecom. We were interested to evaluate software
quality and metrics to know how far our measurement room is inadequate.
The aforementioned experiments (Reverberation Time, Room Impulse Response,
Loudspeaker Characteristics) were repeated with the same procedure as mentioned in
the previous paragraphs. Let us interpret the results :
5.7.1
Reverberations and Impulse Response
Figure 5.11: RT-30 plot inside the dead room
On comparing the results of Figure.5.2 and Figure.5.11 the following points are observed:
36
Figure 5.12: RIR plot inside the dead room
Reverberation time for the dead room is very less (in order of 5 ms) as compared
to actual room (around 22 ms).
There is no evidence of reflections inside the room either in lower frequency or
higher frequency.
Hence, from this observations we conclude that our room had reflections.
As there is less reflections present inside the dead room then possibility of finding
reflections is less useful. However, we find less steep curves in Figure.5.12 as compared
to Figure.5.5.
5.7.2
Loudspeaker Evaluation
From the previous chapters, the frequency response is the single most important aspect
of the performance of any audio device. Basically, it is interesting to consider that for
as long as anyone in acoustics can remember, all electronic devices had basically flat
frequency responses. Yet, no manufacturer of an amplifying device, would momentarily
consider a frequency response specification from some very low frequency to some very
high frequency.
37
Yet, when we come to loudspeakers, it is important to see tolerance of 3 dB or more

are considered acceptable. Hence, we need a mechanism to test the standard deviation
of the loudspeaker for its tolerance.
Hence the following Figure.5.13 for frequency response and standard deviation Figure.5.14
Figure 5.13: Frequency response plot inside dead room
The above frequency response plot shows that the response is flat especially at the
lower frequencies as compared to Figure.5.7. This fact reinforces the problem our room
in lower frequencies where the deviation is in the loudspeakers in the higher frequencies
which can be equalized.
Also, from the Figure 5.14 the deviation from the loudspeaker is within the range
of our tolerance of +/ 3 dB. Yet, at certain frequencies the deviation is high (close
to 3 dB) even in lower frequencies with respect to loudspeaker inside dead room. This
additional information makes us to examine the possibility of inadequate quality of
loudspeaker. Furthermore, this analysis and evaluation will be addressed for effective
design of equalization algorithm.
38
Figure 5.14: Standard Deviation plot for KS Digital loudspeakers
5.8
5.8.1
Proposal for Improvements

Adding Absorbers
From the characterization of the acoustic chamber, one feasible proposal to improve the
bad room character is to add the acoustic absorbers. As stated in the equation 5.1
adding the absorbers is relatively straight forward solution for a room with constant
volume. Hence, the following results stems from adding absorbers in the room.
From Figure.5.15, the absorbers at added at the distinct positions and then room
response is captured in the Figure.5.16. It is clear that absorbers play a crucial role in
reducing the reflections happening inside the room. So without any algorithm compensation this improvement shows a positive sign for room improvement.
Even the reverberation time got reduced (overall) to 8 ms as pictured in the Figure.5.17. Yet, the response with distortions remains same in the lower frequencies.
Adding few absorbers shows a significant improvement in reducing reflections but thats
a one side of the medal the other side is room modes which cant be reduced by adding
absorbers. Yet, suitable best position can be found for best effort equalization of the
Figure 5.15: RIR: room without absorbers
Figure 5.16: RIR: Room improvement with absorbers
39
40
bad room acoustics. Although, complete elimination of room modes is infeasible for this
thesis.
Figure 5.17: RT-30: Room improvement with absorbers
5.9
Summary
To summarize:
Room acoustic analysis consists of three experiments namely Reverberation Time
(RT), Room Impulse Response (RIR) and Room frequency response or Loudspeaker frequency response. In these analysis, distortions present in the non-linear
room frequency response of the acoustic chamber are due to reflections as confirmed by RT and RIR measurements.
Further, to detail why distortions are observed apart from reflections: Room mode
calculations and Room noise analysis are experimented.
Based on the Room acoustic analysis, suitable acoustic absorbers are placed within
the cabin and it significantly reduced reflections. Yet, the other probable distortions caused by either Room modes or Room noise remains unaltered. For all
intents and purposes, we avoid Room modes and Room noise by finding optimal
positions to place loudspeaker and microphone inside the chamber.
41
Hence, the characterization of room acoustics builds suitable notion on how the
cabin behaves acoustically, based on the observed characters a suitable equalization
algorithm will be designed in the next chapter.
Chapter 6
Equalization
The present chapter explains the idea and implementation of equalization algorithm
with integration and verification to the system. The approach briefly describes:
Equalization Design and Implementation: An equalization algorithm is designed

and implemented in a way that the performance of the algorithm is guaranteed
for its efficiency and accuracy. A suitable position is found for the best effort
performance of the algorithm considering bad room effects.
Equalization Verification: Verification explains the statistical accuracy and algorithm run time to guarantee best results. Here, we also discuss the behavior of
algorithm in different cases with correlation to room analysis conducted.
6.1
Equalization Design and Working
This section focuses on the architecture of equalization algorithm in specific which will
be the novelty of this thesis.
6.1.1
Programming Decision
To develop an accurate and efficient equalization algorithm, we asked few questions to

derive our algorithm like:
42
Chapter 6. Equalization
43
How much accuracy can be achieved? In section 5.4, we attempted to have standard test setup to find best positions in the room to avoid reflections. Hence, to
achieve good accuracy we will develop an algorithm for one best position based on
several trails.
How to make the algorithm efficient with minimal run time? We saw in Section
4.1.1, that the SCPI commands are fast enough to communicate with the instrument. Therefore, in order to achieve efficiency, SCPI commands with COM library
will be used to develop the algorithm to have minimal run time.
With this reasoning, we chose to develop the algorithm using these concepts in close
association.
6.1.2
Theory and Block Diagram
The main goal of the algorithm is to reduce bad room characteristics with accuracy of
+/ 0.5 dB and be time efficient(<1 hour). In simple terms, let us say the room boosts
or decreases the amplitude at a specific frequency f 1 by 4 dB, the algorithm we apply
will aim to reduce or increase the offset at that same frequency f 1 by 4 dB.
The Figure.6.1 illustrates the design based on which the equalization algorithm is implemented with equations below.
Let us explain each blocks in detail with a concrete example:
Microphone Calibration: Before we equalize the bad room effect and nonlinearity in the loudspeaker(source), the sensing device microphone needs to be
calibrated. The prime reason is to find the right offset in the microphone. Using
B & K calibrator at 1K Hz(94 dB SPL) generates a value of -40.22 dBV with an
accuracy of +/ 0.2 dB. Hence, the total offset from the microphone will be 134
dB which needs to be added with measured value from the microphone.
Volume Adjustments: Volume adjustments aims to switch on the source and
record a value from the microphone for a tone at 1k Hz. Based on the recorded
value, now it adjusts the volume to the defined target sound pressure level (SPL)
44
Figure 6.1: Equalization algorithm design: adaptation from M.Sc seminar
say 90 dB or any value of interest. The entire function completes the calibration
with an accuracy of +/ 0.02 dB. The new voltage is found using equation 6.2.
To find initial voltage say xdBV correction at 1k Hz for a target SPL of 90 dB
SPL(-43 dBV) the following relationship is used :
xdBV = 20 log10 (V /V0 )
(6.1)
Where V is the correction voltage and V0 is reference voltage normally 1 V.

Rearranging the above equation yields,
V = 10
xdBV
20
(6.2)
The above equation 6.2 is run iteratively to find the actual voltage. For our
example at 90dB SPL the applied voltage is 69.7 mV(millivolt) which is to be
applied to all frequencies is filled into column vector:
Vnormal = V I
(6.3)
45
here I is identity matrix
Vnormal
69.7
69.7
= ..
..
69.7
(6.4)
Sweep and Store Values: Based on applied correction voltage from equation
6.3, a frequency sweep F req is made and the output response is:
200
205.67
F req = ..
..
10, 000
Xuneq
41.38
41.39
= ..
..
33.24
(6.5)
(6.6)
Where, Xuneq is th output in dBV and F req is the frequency points from 200-10k
Hz consisting of 150 data points.
First Equalization: Based on the frequency sweep at the defined target sound
pressure level, the first equalization finds a offset factor as given in equation 6.7.
To find offset between the target SPL and measured value (Xuneq ), we derive an
offset factor KF actor as using below:
KF actor (i) 10
targetSP LXuneq (i)

20
(6.7)
For our first iteration, target SPL = -43.0 and Xuneq = -41.38 (for 200 Hz) on
plugging in the equation 6.7 we get:
46
KF actor
0.7784
0.8114
= ..
..
0.3175
(6.8)
So, first equalized value Eequal is generated as:

EEqual = Vnormal KF actor
(6.9)
The above equation produces right compensation for the bad room acoustics during equalization. So, the first equalization means applying the equation 6.9 and
equalizing the response from the system.
Check Accuracy and Second Equalization: Further, we check the accuracy
and if its not as desired (+/ 0.5 dB) then voltage is further rounded off for more
precision and algorithm runs in second iteration as termed as second equalization
System Halt: Finally, after second equalization, system is switched off. The
obtained data is processed and values are stored.
A self-explanatory flowchart is illustrated in Figure.6.2
6.2
Setup and Results
Figure.6.3 shows the best case arrangement for the the room before equalization to avoid
room modes and acoustic noise inside the room. Here, this position is found in way that
it is close to (2/3)rd away from the mirrored wall in breadth. It is also (1/2)th in the
length. Based on several trail and error methods this position is found. Hence, on this
position algorithm works stably with guaranteed accuracy.
Figure.6.4 shows the output equalized response from the system with respect to 90
and 80 dB SPL. As we see from the figure the whole response is equalized at target SPL.
The accuracy of the overall algorithm with respect to 90 and 80 dB is +/ 0.2 dB for
all frequencies. The another important fact is that the output response is not spurious
and is close to a straight line at the target SPL.
Figure 6.2: Equalization module flow chart
47
48
Figure 6.3: Best case setup inside acoustic chamber
For the target SPL of 70 and 60 dB SPL the output is indicated in the Figure.6.5.
For 70 dB SPL the outputs behaves similar to 90 and 80 dB SPL. Yet, for 60 dB SPL
the deviation is up to than +/ 0.2 dB. Also, the curves gets little spurious in the lower
frequency region. However, the curve looks straight overall with satisfying accuracy.
From the Figure.6.6, the curve is not straight or ideal as per the requirement. However, the output is equalized at the target SPL which is 50 dB SPL and is within the
range of +/ 0.5 dB SPL.
Main reasons for not getting ideal output response curve is :
Influence of noise level at lower sound pressure levels

Room Modes at lower frequencies
The above two mentioned points were already concluded in the Chapter 5. However, we
will do detailed statistical analysis in the below sections.
49
Figure 6.4: Equalization results for 90 and 80 dB SPL
6.3
Verification
To verify the accuracy of our algorithm, we conducted two experiments with a goal to
test the algorithm:
6.3.1
Verification at 50 dB SPL
Here the algorithm is tested at 50 dB SPL where the same correction factor as given in
equation.6.8 is applied in two trails without changing or modifying values. As per the
definition of testing, we should observe a repeatable results again. Yet, the results are not
same which reasons the inadequacy of the algorithm at 50 dB SPL. As stated, presence
of room modes at lower frequencies makes the system immune to applied voltage. Also,
Figure 6.5: Equalization results for 70 and 60 dB SPL
Figure 6.6: Equalization results for 50 dB SPL
50
51
the points where the deviation is higher than 0.5 dB are similar points or frequencies as
observed in the noise response from the room around 200-400 Hz illustrated in Figure.6.7.
Although, the verification between two trails is not 100 % accurate, yet our algorithm
does better by leaving 1% error out of 150 data points which is still acceptable.
Figure 6.7: Verification at 50 dB SPL
6.3.2
Statistical Accuracy at 50 dB SPL
To find the statistical accuracies at the 50 dB SPL, the equalization algorithm is executed
several trails(approx.25) to observe the deviation in the equalized points as shown in
Figure.6.8. Every time the correction voltage is changed and same voltage is not applied
again. This experimental procedure gave results conforming the deviation in accuracy
up to +/ 0.5 dB specially at lower frequencies(less than 1000 Hz). This experiment
again proves that our solution is still acceptable in terms of accuracy at lower sound
pressure levels (50 dB).
52
Figure 6.8: Statistical Deviation with +/ 0.5 dB
6.4
Algorithm Runtime
Having accurate results from the equalization algorithm is one side of the medal, another
side is to be time efficient in equalizing the values within the defined time-budget (less
than 1 hour). To see, how far our developed algorithm is time efficient, a comparison
bar plot is made as illustrated in the Figure.6.9. From the diagram, the algorithm
runs for about 10 minutes to equalize 150 data points which makes our algorithm more
efficient. Similarly, with increase in the samples or data points the run time increases
proportionally. Yet, in all the cases the run time is well within the time-budget. The
run time computed here is from start of the system with user intervention and actual
equalization completed.
6.5
System Testing
Here the utilization of the equalization algorithm is done by the system to evaluate
the performance of the hearing aid algorithms using default R & S UPV software and
implemented novel equalization algorithm.
53
Figure 6.9: Algorithm run time as size of data increases
Figure 6.10: Field of frequency response: using R &S system
6.5.1
Comparison of Field Frequency Response
The field of frequency response is the frequency sweep algorithm done in the hearing
aid at various SPL. The algorithm runs from the lowest sound pressure level to the
highest sound pressure(50 dB SPL to 90 dB SPL) spanning from 100-10k Hz. This
complete algorithm is offered in UPV-K7 which results in the output response as shown
in the Figure.6.10 . As plotted in the figure the algorithm does not produce accurate
results as per the wish-list which is +/ 0.5 dB. The algorithm leaves two or many
points deviated.
54
Figure 6.11: Field of frequency response: using novel equalization algorithm
To avoid this problem, our novel equalization algorithm is verified for the system at
various SPL. The results are illustrated in the Figure.6.11. As we compare the results
from default R & S UPV equalization algorithm to our new equalization algorithm, definitely the new proposed algorithm outperforms the proprietary solutions with increased
accuracy and with increased efficiency satisfying IEC standards.
6.6
Summary
To summarize:
We developed a complete novel equalization algorithm based on our understanding

of room acoustics by suitable characterization measurements inside the chamber.
The possible distortions are compensated by finding suitable positions to place
loudspeaker and microphones inside the chamber.
The developed algorithm satisfies the IEC standard requirements at all sound
pressure levels. Yet, few points (less than 1% of all data points) may deviate more
than +/ 0.5 dB at 50 dB SPL which is still adequate for hearing aids testing.
System testing further verifies our eminence of our developed algorithm as it significantly outperforms the existing solutions (both R & S K-7 and Cetecom solutions).
Chapter 7
Conclusion and Future Work

The present Master thesis implements an automated remote control based solutions
to test and measure electro-acoustical parameters in hearing aids by developing novel
equalization algorithm to compensate unfavorable room effects. In particular, the thesis
presents detailed analysis of the room characteristics with possible improvements to the
room acoustics. Based on the technical analysis, a suitable equalization algorithm is
designed and implemented so as to satisfy the performance requirements introduced
by IEC 60118 standard for hearing aids measurements and discusses the ability of the
developed algorithm to meet the requirements in terms of accuracy and time efficiency.
The characterization of the room acoustics is well explained by techniques namely:
Reverberation Time, Room Impulse Response in the acoustic chamber which indicated
the presence of reflections inside the room. By adding few acoustic absorbers the reflections inside the room got compensated. However, the distortions in the frequency
response of the room is not completely compensated by absorbers as it is caused by
the presence of room modes and acoustic noise at lower frequencies and lower sound
pressure levels. Practically, we tried to avoid room modes and acoustic noise by placing
the acoustic components optimally inside the chamber during equalization. Hence, the
characterization shows that with suitable absorbers the room response can be improved
significantly.
The novel equalization algorithm developed from the room characterization with
optimal positions shows a significant improvement in the accuracy and time efficiency
as confirmed by system verification and statistical analysis. The performance evaluation
55
Chapter 7. Conclusion and Future Work
56
conducted between the new equalized algorithm and default proprietary solutions indicates that the new algorithm outperforms default algorithm in terms of accuracy and
efficiency. Although, the developed solution is not completely accurate in lower sound
pressure levels such as 50 dB. Still, our solution produces a reliable and acceptable results in terms of accuracy +/ 0.5 dB and run time efficiency within 1 hour which forms
the basis to satisfy IEC standards for hearing aids compliance testing.
We claim novelty in the thesis for having a stringent requirements to develop
equalization algorithm with +/ 0.5 dB which runs within 1 hour. Throughout our
work, we had one crucial challenge as to develop an equalization algorithm without using
any post-processing of signals which makes our design and implementation unique. Most
importantly, our developed solution accomplishes an acceptable and reliable results at all
sound pressure levels satisfying strict requirements from IEC standards which elevates
novelty by all measures.
The thesis presents a technically feasible implementation to measure electro-acoustical
entities of hearing aids by systematic analysis of the room acoustics by developing a
novel self-calibrating algorithm to equalize unfavorable effects of bad room acoustics.
The satisfaction of performance metrics such as accuracy and time efficiency set by the
IEC standards elevates the capability of our solution for the next-generation test and
measurement systems in the context of electro-acoustical measurements. Though, the
developed solutions concentrates on hearing aids measurements yet the automated solutions constitute a vibrant research and development field especially in the area of room
acoustics. Hence, a variety of technical challenges are therefore awaiting solutions.
There is always room for improvement. The evaluation of the equalization algorithm
at lower sound pressure levels and lower frequencies brings up new ideas and challenges,
One feasible implementation involving room modal based compensation focusing on 50
dB SPL can be realized. However, before developing modules on equalization several
frequency response experiments with new loudspeakers can be done as a quantitative
measures as we doubt the performance of existing loudspeaker. Similarly, an effective
characterization of the sound system amplifiers needs to be conducted as a possible
future work.
Bibliography
[1] Michael Valente. Hearing aids: Standards, options, and limitations. Thieme, 2002.
[2] James Jerger. Behavioral correlates of hearing-aid performance. Bull. Prosthetics
Res. Spring, 1967.
[3] John N Mourjopoulos. Digital equalization of room acoustics. Journal of the Audio
Engineering Society, 42(11):884900, 1994.
[4] Nikolai Bisgaard, Marcel SMG Vlaming, and Martin Dahlquist. Standard audiograms for the iec 60118-15 measurement procedure. Trends in amplification, 14(2):
113120, 2010.
[5] Matti Karjalainen, Esa Piirila, Antti Jarvinen, and Jyri Huopaniemi. Comparison
of loudspeaker equalization methods based on dsp techniques. Journal of the Audio
Engineering Society, 47(1/2):1431, 1999.
[6] Aki M
akivirta, Poju Antsalo, Matti Karjalainen, and Vesa Valimaki. Modal equalization of loudspeaker-room responses at low frequencies. Journal of the Audio
Engineering Society, 51(5):324343, 2003.
[7] Arie JM Kaizer. Modeling of the nonlinear response of an electrodynamic loudspeaker by a volterra series expansion. Journal of the Audio Engineering Society,
35(6):421433, 1987.
[8] Wolfgang Klippel and Robert Werner. Loudspeaker distortionmeasurement and
perception.
[9] SCPI Consortium et al. Standard commands for programmable instruments, 1999.
57
Bibliography
58
[10] Dany Cheij. A software architecture for building interchangeable test systems. In
AUTOTESTCON Proceedings, 2001. IEEE Systems Readiness Technology Conference, pages 1622. IEEE, 2001.
[11] Steve Burbeck. Applications programming in smalltalk-80 (tm): How to use modelview-controller (mvc). Smalltalk-80 v2, 5, 1992.
[12] Ant
onio Pedro O Carvalho. The use of the sabine and eyring reverberation time
equations to churches. The Journal of the Acoustical Society of America, 97(5):
33193319, 1995.
[13] John S Bradley. Using iso 3382 measures, and their extensions, to evaluate acoustical conditions in concert halls. Acoustical science and technology, 26(2):170178,
2005.
[14] Angelo Farina. Advancements in impulse response measurements by sine sweeps.
In Audio Engineering Society Convention 122. Audio Engineering Society, 2007.
[15] Jukka P
atynen, Brian FG Katz, and Tapio Lokki. Investigations on the balloon
as an impulse source. The Journal of the Acoustical Society of America, 129(1):
EL27EL33, 2011.
[16] Floyd E Toole. Loudspeakers and rooms for sound reproductiona scientific review.
Journal of the audio engineering society, 54(6):451476, 2006.
[17] Heinrich Kuttruff. Room acoustics. CRC Press, 2009.
[18] Swen M
uller and Paulo Massarani. Transfer-function measurement with sweeps.
Journal of the Audio Engineering Society, 49(6):443471, 2001.
[19] Thomas D Rossing. Springer handbook of acoustics. Springer, 2007.
[20] Oscar J Bonello. A new criterion for the distribution of normal room modes. Journal
of the Audio Engineering Society, 29(9):597606, 1981.

Acoustics

Enviado por

Dados do documento

Título original

Direitos autorais

Formatos disponíveis

Compartilhar este documento

Compartilhar ou incorporar documento

Opções de compartilhamento

Você considera este documento útil?

Este conteúdo é inapropriado?

Direitos autorais:

Formatos disponíveis

Acoustics

Enviado por

Direitos autorais:

Formatos disponíveis

Universit

Development of Test and Measurement

Submitted to Saarland University

Mohamed Imran Noor Mohamed

Prof. Dr. Dietrich Klakow

Dipl.-Ing. Oleg Fallmann

Prof. Dr. Dietrich Klakow

Statement in Lieu of an Oath

(Ort und Datum / Place and Date)

(Ort und Datum / Place and Date)

5 Characterization of Room Acoustics

7 Conclusion and Future Work

Sound source calibration by R & S; K-7 software . . . . . . . . . . . . . .

Condensation and rarefaction effect of sound . . . . . . . . . . . . . . . . 11

Example SCPI command explaining device operation . . . .

Flow chart explaining RT30 measurement . . . . . . . . .

Equalization algorithm design: adaptation from

6.9 Algorithm run time as size of data increases . . . . . . . . . . . . . . . . . 53

International Organization for Standardization

International Electrotechnical Commission

Sound Pressure Level

Bruel and Kaer

Rhode and Schwarz

Standard Commands Programmable Instruments

Interchangeable Virtual Instruments

Virtual Instruments Software Architecture

Model View Controller

Digital Signal Processing

Room Impulse Response

Need for Calibration

Chapter 2. Problem Formulation

Measurement of the absolute sensitivity at one frequency

Chapter 2. Problem Formulation

Equalization by R&S K-7 Software

Figure 2.1: Sound source calibration by R & S; K-7 software

As an illustration, we did equalization for the sound source (loudspeaker) on sound

Chapter 2. Problem Formulation

Equalization by Cetecom software

To solve the equalization problem, Cetecom developed an algorithm to linearize the

Figure 2.2: Sound source calibration by Cetecom(Minimum Linearization)

Chapter 2. Problem Formulation

Figure 2.3: Sound source calibration by Cetecom(Maximum Linearization)

Chapter 2. Problem Formulation

Chapter 3. Acoustics: Overview

Figure 3.1: Condensation and rarefaction effect of sound

Chapter 3. Acoustics: Overview

Sound Level and the Decibel

As compared to other physical units and measurements in acoustics, the intensity of

I(dB) = 10 log10 [I/I0 ]

Where I0 = 1012 W atts/m2 ,

L(dBV ) = 20 log10 (v/v0 )

Chapter 3. Acoustics: Overview

A microphone is an acoustics-to-electric transducer which converts the sound pressure

Chapter 3. Acoustics: Overview

Figure 3.2: Sound travel inside a room

The phenomenon of having reflected sounds in an enclosure is known as Reverberation.

Chapter 3. Acoustics: Overview

Software Design and Architecture

Figure 4.1: Example SCPI command explaining device operation

Virtual Instrument Software Architecture, commonly known as VISA, is a widely used

Figure 4.2: VISA hierarchy with different communication bus protocols

Finally, it provides a unified API to communicate between test equipment-regardless

Hence, we employed the VISA- software architecture to communicate between our