Abstract: Noise disturbance is one of the major factors considered in the fast development of aircraft technology. This paper reviews the flow field, which is examined on the 2D NACA0015 and 3D NACA0012 blade profile using SST k-ω turbulence model to compute the unsteady flow field. We inserted the time-dependent flow area variables in Ffowcs-Williams and Hawkings (FW-H) equations as an input and Sound Pressure Level (SPL) values will be computed for different angles of attack (AoA) from the microphone which is positioned in the computational domain to investigate effect of augmentation of unsteady 2D and 3D airfoil region noise level. The computed results will be compared with experimental data which are available in the open literature. As results; one of the calculated Cp is slightly lower than the experimental value. This difference could be due to the higher Reynolds number of the experimental data. The ANSYS Fluent software was used in this study. Fluent includes well-validated physical modeling capabilities to deliver fast, accurate results across the widest range of CFD and multiphysics applications. This paper includes a study which is on external flow over an airfoil. The case of 2D NACA0015 has approximately 7 million elements and solves compressible fluid flow with heat transfer using the SST turbulence model. The other case of 3D NACA0012 has approximately 3 million elements.
Abstract: For many years, the ear protectors have been used to preventing the audio and non-audio effects of received noise from occupation environments. Despite performing hearing protection programs, there are many people which still suffer from noise-induced hearing loss. This study was conducted with the aim of determination of human hearing system response to received noise and the effectiveness of ear protectors on preventing of noise-induced hearing loss. Sound pressure microphones were placed in a simulated ear canal. The severity of noise measured inside and outside of ear canal. The noise reduction values due to installing ear protectors were calculated in the octave band frequencies and LabVIEW programmer. The results of noise measurement inside and outside of ear canal showed a different in received sound levels by ear canal. The effectiveness of ear protectors has been considerably reduced for the low frequency limits. A change in resonance frequency also was observed after using ear protectors. The study indicated the ear canal structure may affect the received noise and it may lead a difference between the received sound from the measured sound by a sound level meter, and hearing system. It means the human hearing system may probably respond different from a sound level meter. Hearing protectors’ efficiency declines by increasing the noise levels, and thus, they are not suitable to protect workers against industrial noise particularly low frequency noise. Hearing protectors may be solely a reason to damaging of hearing system in a special frequency via changing of human hearing system acoustical structure. We need developing the subjective method of hearing protectors testing, because their evaluation is not designed based on industrial noise or in the field.
Abstract: A 3C-2D PIV technique was applied to investigate the swirling flow generated by an axial plus tangential type swirl generator. This work is focused on the near-exit region of an isothermal swirling jet to characterize the effect of swirl on the flow field and to identify the large coherent structures both in unconfined and confined conditions for geometrical swirl number, Sg = 4.6. Effects of the Reynolds number on the flow structure were also studied. The experimental results show significant effects of the confinement on the mean velocity fields and its fluctuations. The size of the recirculation zone was significantly enlarged upon confinement compared to the free swirling jet. Increasing in the Reynolds number further enhanced the recirculation zone. The frequency characteristics have been measured with a capacitive microphone which indicates the presence of periodic oscillation related to the existence of precessing vortex core, PVC. Proper orthogonal decomposition of the jet velocity field was carried out, enabling the identification of coherent structures. The time coefficients of the first two most energetic POD modes were used to reconstruct the phase-averaged velocity field of the oscillatory motion in the swirling flow. The instantaneous minima of negative swirl strength values calculated from the instantaneous velocity field revealed the presence of two helical structures located in the inner and outer shear layers and this structure fade out at an axial location of approximately z/D = 1.5 for unconfined case and z/D = 1.2 for confined case. By phase averaging the instantaneous swirling strength maps, the 3D helical vortex structure was reconstructed.
Abstract: New sensors and technologies – such as microphones,
touchscreens or infrared sensors – are currently making their
appearance in the automotive sector, introducing new kinds of
Human-Machine Interfaces (HMIs). The interactions with such tools
might be cognitively expensive, thus unsuitable for driving tasks.
It could for instance be dangerous to use touchscreens with a
visual feedback while driving, as it distracts the driver’s visual
attention away from the road. Furthermore, new technologies in
car cockpits modify the interactions of the users with the central
system. In particular, touchscreens are preferred to arrays of buttons
for space improvement and design purposes. However, the buttons’
tactile feedback is no more available to the driver, which makes
such interfaces more difficult to manipulate while driving. Gestures
combined with an auditory feedback might therefore constitute an
interesting alternative to interact with the HMI. Indeed, gestures can
be performed without vision, which means that the driver’s visual
attention can be totally dedicated to the driving task. In fact, the
auditory feedback can both inform the driver with respect to the task
performed on the interface and on the performed gesture, which might
constitute a possible solution to the lack of tactile information. As
audition is a relatively unused sense in automotive contexts, gesture
sonification can contribute to reducing the cognitive load thanks
to the proposed multisensory exploitation. Our approach consists
in using a virtual object (VO) to sonify the consequences of the
gesture rather than the gesture itself. This approach is motivated
by an ecological point of view: Gestures do not make sound, but
their consequences do. In this experiment, the aim was to identify
efficient sound strategies, to transmit dynamic information of VOs to
users through sound. The swipe gesture was chosen for this purpose,
as it is commonly used in current and new interfaces. We chose
two VO parameters to sonify, the hand-VO distance and the VO
velocity. Two kinds of sound parameters can be chosen to sonify the
VO behavior: Spectral or temporal parameters. Pitch and brightness
were tested as spectral parameters, and amplitude modulation as a
temporal parameter. Performances showed a positive effect of sound
compared to a no-sound situation, revealing the usefulness of sounds
to accomplish the task.
Abstract: This paper investigates the characteristics of wall
pressure fluctuations in naturally developing boundary layer flows
on axisymmetric bodies experimentally. The axisymmetric body has
a modified ellipsoidal blunt nose. Flush-mounted microphones are
used to measure the wall pressure fluctuations in the boundary layer
flow over the body. The measurements are performed in a low noise
wind tunnel. It is found that the correlation between the flow regime
and the characteristics of the pressure fluctuations is distinct. The
process from small fluctuation in laminar flow to large fluctuation in
turbulent flow is investigated. Tollmien-Schlichting wave (T-S wave)
is found to generate and develop in transition. Because of the T-S
wave, the wall pressure fluctuations in the transition region are higher
than those in the turbulent boundary layer.
Abstract: This paper describes a novel sensor device, a pressure
pulse wave meter, which uses a bidirectional condenser microphone.
The microphone work as a microphone as well as a sensor with high
gain over a wide frequency range; they are also highly reliable and
economic. Currently aging is becoming a serious social issue in Japan causing
increased medical expenses in the country. Hence, it is important for
elderly citizens to check health condition at home, and to care the
health conditions through daily monitoring. Given this circumstances,
we developed a novel pressure pulse wave meter based on a
bidirectional condenser microphone: this device is used as a measuring
instrument of health conditions.
Abstract: Pipa is one of the most important Chinese traditional
plucked instruments, but its directivity has never been measured
systematically. In western, directivity of loudness for western
instruments is deeply researched through analysis of sound pressure
level, whereas the directivity of timbre is seldom studied. In this paper,
a new method for directivity of timbre was proposed, and horizontal
directivity patterns of loudness and timbre of Pipa were measured.
Directivity of Pipa radiation was measured in an anechoic room. The
sound of Pipa played by a musician was recorded simultaneously by
32 microphones with Pipa in the center. The measuring results were
examined through listening test. According to the measurement of
Pipa directivity radiation, we put forward the best localization of Pipa
in the Chinese traditional orchestra and the optimal recording region.
Abstract: In this work, a method of time delay estimation for
dual-channel acoustic signals (speech, music, etc.) recorded under
reverberant conditions is investigated. Standard methods based on
cross-correlation of the signals show poor results in cases involving
strong reverberation, large distances between microphones and
asynchronous recordings. Under similar conditions, a method based
on cross-correlation of temporal envelopes of the signals delivers a
delay estimation of acceptable quality. This method and its properties
are described and investigated in detail, including its limits of
applicability. The method’s optimal parameter estimation and a
comparison with other known methods of time delay estimation are
also provided.
Abstract: Earphones and headphones, which are compact electro-acoustic transducers, tend to have a lot of acoustic absorption materials and porous materials known as dampers, which often have a large number of extremely small holes and narrow slits to inhibit the resonance of the vibrating system, because the air viscosity significantly affects the acoustic characteristics in such acoustic paths. In order to perform simulations using the finite element method (FEM), it is necessary to be aware of material characteristics such as the impedance and propagation constants of sound absorbing materials and porous materials. The transfer function is widely known as a measurement method for an acoustic tube with such physical properties, but literature describing the measurements at the upper limits of the audible range is yet to be found. The acoustic tube, which is a measurement instrument, must be made narrow, and the distance between the two sets of microphones must be shortened in order to take measurements of acoustic characteristics at higher frequencies. When such a tube is made narrow, however, the characteristic impedance has been observed to become lower than the impedance of air. This paper considers the cause of this phenomenon to be the effect of the air viscosity and describes an FEM analysis of an acoustic tube considering air viscosity to compare to the theoretical formula by including the effect of air viscosity in the theoretical formula for an acoustic tube.
Abstract: Headphones and earphones have many extremely small
holes or narrow slits; they use sound-absorbing or porous material (i.e.,
dampers) to suppress vibratory system resonance. The air viscosity in
these acoustic paths greatly affects the acoustic properties. Simulation
analyses such as the finite element method (FEM) therefore require
knowledge of the material properties of sound-absorbing or porous
materials, such as the characteristic impedance and propagation
constant. The transfer function method using acoustic tubes is a widely
known measuring method, but there is no literature on taking
measurements up to the audible range. To measure the acoustic
properties at high-range frequencies, the acoustic tubes that form the
measuring device need to be narrowed, and the distance between the
two microphones needs to be reduced. However, when the tubes are
narrowed, the characteristic impedance drops below the air impedance.
In this study, we considered the effect of air viscosity in an acoustical
tube, introduced a theoretical formula for this effect in the form of
complex density and complex sonic velocity, and verified the
theoretical formula. We also conducted an experiment and observed
the effect from air viscosity in the actual measurements.
Abstract: In this study, noise characteristics of structure were analyzed in an effort to reduce noise passing through an opening of an
enclosure surrounding the structure that generates noise. Enclosures
are essential measure to protect noise propagation from operating machinery. Access openings of the enclosures are important path of noise leakage. First, noise characteristics of structure were analyzed
and feed-forward noise control was performed using simulation in
order to reduce noise passing through the opening of enclosure, which
surrounds a structure generating noise. We then implemented a
feed-forward controller to actively control the acoustic power through
the opening. Finally, we conducted optimization of placement of the
reference sensors for several cases of the number of sensors. Good
control performances were achieved using the minimum number of microphones arranged an optimal placement.
Abstract: Frequency domain independent component analysis has
a scaling indeterminacy and a permutation problem. The scaling
indeterminacy can be solved by use of a decomposed spectrum. For
the permutation problem, we have proposed the rules in terms of gain
ratio and phase difference derived from the decomposed spectra and
the source-s coarse directions.
The present paper experimentally clarifies that the gain ratio and
the phase difference work effectively in a real environment but their
performance depends on frequency bands, a microphone-space and
a source-microphone distance. From these facts it is seen that it is
difficult to attain a perfect solution for the permutation problem in a
real environment only by either the gain ratio or the phase difference.
For the perfect solution, this paper gives a solution to the problems
in a real environment. The proposed method is simple, the amount of
calculation is small. And the method has high correction performance
without depending on the frequency bands and distances from source
signals to microphones. Furthermore, it can be applied under the real
environment. From several experiments in a real room, it clarifies
that the proposed method has been verified.
Abstract: Monitoring the tool flank wear without affecting the
throughput is considered as the prudent method in production
technology. The examination has to be done without affecting the
machining process. In this paper we proposed a novel work that is
used to determine tool flank wear by observing the sound signals
emitted during the turning process. The work-piece material we used
here is steel and aluminum and the cutting insert was carbide
material. Two different cutting speeds were used in this work. The
feed rate and the cutting depth were constant whereas the flank wear
was a variable. The emitted sound signal of a fresh tool (0 mm flank
wear) a slightly worn tool (0.2 -0.25 mm flank wear) and a severely
worn tool (0.4mm and above flank wear) during turning process were
recorded separately using a high sensitive microphone. Analysis
using Singular Value Decomposition was done on these sound
signals to extract the feature sound components. Observation of the
results showed that an increase in tool flank wear correlates with an
increase in the values of SVD features produced out of the sound
signals for both the materials. Hence it can be concluded that wear
monitoring of tool flank during turning process using SVD features
with the Fuzzy C means classification on the emitted sound signal is
a potential and relatively simple method.
Abstract: A set of Artificial Neural Network (ANN) based methods
for the design of an effective system of speech recognition of
numerals of Assamese language captured under varied recording
conditions and moods is presented here. The work is related to
the formulation of several ANN models configured to use Linear
Predictive Code (LPC), Principal Component Analysis (PCA) and
other features to tackle mood and gender variations uttering numbers
as part of an Automatic Speech Recognition (ASR) system in
Assamese. The ANN models are designed using a combination of
Self Organizing Map (SOM) and Multi Layer Perceptron (MLP)
constituting a Learning Vector Quantization (LVQ) block trained in a
cooperative environment to handle male and female speech samples
of numerals of Assamese- a language spoken by a sizable population
in the North-Eastern part of India. The work provides a comparative
evaluation of several such combinations while subjected to handle
speech samples with gender based differences captured by a microphone
in four different conditions viz. noiseless, noise mixed, stressed
and stress-free.
Abstract: A combined three-microphone voice activity detector (VAD) and noise-canceling system is studied to enhance speech recognition in an automobile environment. A previous experiment clearly shows the ability of the composite system to cancel a single noise source outside of a defined zone. This paper investigates the performance of the composite system when there are frequently moving noise sources (noise sources are coming from different locations but are not always presented at the same time) e.g. there is other passenger speech or speech from a radio when a desired speech is presented. To work in a frequently moving noise sources environment, whilst a three-microphone voice activity detector (VAD) detects voice from a “VAD valid zone", the 3-microphone noise canceller uses a “noise canceller valid zone" defined in freespace around the users head. Therefore, a desired voice should be in the intersection of the noise canceller valid zone and VAD valid zone. Thus all noise is suppressed outside this intersection of area. Experiments are shown for a real environment e.g. all results were recorded in a car by omni-directional electret condenser microphones.
Abstract: An investigation of noise in a micro stepping motor is
considered to study in this article. Because of the trend towards higher
precision and more and more small 3C (including Computer,
Communication and Consumer Electronics) products, the micro
stepping motor is frequently used to drive the micro system or the
other 3C products. Unfortunately, noise in a micro stepped motor is
too large to accept by the customs. To depress the noise of a micro
stepped motor, the dynamic characteristics in this system must be
studied. In this article, a Visual Basic (VB) computer program speed
controlled micro stepped motor in a digital camera is investigated.
Karman KD2300-2S non-contract eddy current displacement sensor,
probe microphone, and HP 35670A analyzer are employed to analyze
the dynamic characteristics of vibration and noise in a motor. The
vibration and noise measurement of different type of bearings and
different treatment of coils are compared. The rotating components,
bearings, coil, etc. of the motor play the important roles in producing
vibration and noise. It is found that the noise will be depressed about
3~4 dB and 6~7 dB, when substitutes the copper bearing with plastic
one and coats the motor coil with paraffin wax, respectively.
Abstract: This paper presents how the real-time chatter
prevention can be realized by feedback of acoustic cutting signal, and
the efficacy of the proposed adaptive spindle speed tuning algorithm is
verified by intensive experimental simulations. A pair of
microphones, perpendicular to each other, is used to acquire the
acoustic cutting signal resulting from milling chatter. A real-time
feedback control loop is constructed for spindle speed compensation
so that the milling process can be ensured to be within the stability
zone of stability lobe diagram. Acoustic Chatter Signal Index (ACSI)
and Spindle Speed Compensation Strategy (SSCS) are proposed to
quantify the acoustic signal and actively tune the spindle speed
respectively. By converting the acoustic feedback signal into ACSI,
an appropriate Spindle Speed Compensation Rate (SSCR) can be
determined by SSCS based on real-time chatter level or ACSI.
Accordingly, the compensation command, referred to as Added-On
Voltage (AOV), is applied to increase/decrease the spindle motor
speed. By inspection on the precision and quality of the workpiece
surface after milling, the efficacy of the real-time chatter prevention
strategy via acoustic signal feedback is further assured.
Abstract: A state of the art Speaker Identification (SI) system
requires a robust feature extraction unit followed by a speaker
modeling scheme for generalized representation of these features.
Over the years, Mel-Frequency Cepstral Coefficients (MFCC)
modeled on the human auditory system has been used as a standard
acoustic feature set for speech related applications. On a recent
contribution by authors, it has been shown that the Inverted Mel-
Frequency Cepstral Coefficients (IMFCC) is useful feature set for
SI, which contains complementary information present in high
frequency region. This paper introduces the Gaussian shaped filter
(GF) while calculating MFCC and IMFCC in place of typical
triangular shaped bins. The objective is to introduce a higher
amount of correlation between subband outputs. The performances
of both MFCC & IMFCC improve with GF over conventional
triangular filter (TF) based implementation, individually as well as
in combination. With GMM as speaker modeling paradigm, the
performances of proposed GF based MFCC and IMFCC in
individual and fused mode have been verified in two standard
databases YOHO, (Microphone Speech) and POLYCOST
(Telephone Speech) each of which has more than 130 speakers.
Abstract: Revolutions Applications such as telecommunications, hands-free communications, recording, etc. which need at least one microphone, the signal is usually infected by noise and echo. The important application is the speech enhancement, which is done to remove suppressed noises and echoes taken by a microphone, beside preferred speech. Accordingly, the microphone signal has to be cleaned using digital signal processing DSP tools before it is played out, transmitted, or stored. Engineers have so far tried different approaches to improving the speech by get back the desired speech signal from the noisy observations. Especially Mobile communication, so in this paper will do reconstruction of the speech signal, observed in additive background noise, using the Kalman filter technique to estimate the parameters of the Autoregressive Process (AR) in the state space model and the output speech signal obtained by the MATLAB. The accurate estimation by Kalman filter on speech would enhance and reduce the noise then compare and discuss the results between actual values and estimated values which produce the reconstructed signals.
Abstract: Water leakage is a serious problem in the maintenance of a waterworks facility. Monitoring the water flow rate is one way to locate leakage. However, conventional flowmeters such as the wet-type flowmeter and the clamp-on type ultrasonic flowmeter require additional construction for their installation and are therefore quite expensive. This paper proposes a novel estimation system for the flow rate in a water pipeline, which employs a vibration sensor. This assembly can be attached to any water pipeline without the need for additional high-cost construction. The vibration sensor is designed based on a condenser microphone. This sensor detects vibration caused by water flowing through a pipeline. It is possible to estimate the water flow rate by measuring the amplitude of the output signal from the vibration sensor. We confirmed the validity of the proposed sensing system experimentally.