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Department Library

2019

Chad Clayton (Capstone, August 2019, Advisor: Scott Sommerfeldt )

Abstract

The Blendtec Total Home Blender is one of the most popular high-powered home blenders in America and the most consistent customer complaint since its release has been the noise. It is extremely loud. I own one of the blenders and have often wondered how it could be made quieter. A number of years ago, Blendtec contacted BYU Acoustics and asked them for help developing noise-cancelling technology for their commercial blenders. Their solution, a large plastic enclosure, can now be seen at popular smoothie shops like Jamba Juice that use commercial blenders. But no similar success has been found with Blendtec’s retail blenders designed for use in the household. I met with Blendtec’s Head of Engineering, David Throckmorton, and learned that challenges related to price, operating space, and legal marketability have prevented Blendtec from developing effective damping for their Total Home Blender (their most popular product). I worked with David to narrow down some design ideas which I then took to Dr. Scott Sommerfeldt. The constraints of the retail project required a design that was low-cost, unobtrusive, and legal for retail sale. Together we decided to try two separate solutions: ribbing on the blender jar to prevent it from functioning as a loudspeaker and vibration-absorbing feet for the blender base to sit in to prevent noise from vibration against the countertop. I designed and constructed both apparatuses and took measurements in BYU’s anechoic chamber using LabView (AFR) and analyzing the data in MatLab. The ribbed jar was completely ineffective but the vibration-absorbing feet showed some promise, with the best configuration showing an average reduction of ~4dB across audible frequencies. All materials were then turned in, along with the findings, to Blendtec for their use in future design.

Caleb Burley Goates (Masters Thesis, June 2019, Advisor: Scott Sommerfeldt )

Abstract

The search for a convenient connection between vibration patterns on a structure and the sound radiated from that structure is ongoing in structural acoustics literature. Common techniques are wavenumber domain methods, or representation of the vibration in terms of some basis, such as structural modes or elementary radiators, and calculating the sound radiation in terms of the basis. Most choices for a basis in this situation exhibit strong coupling between the basis functions, but there is one choice which does not: Acoustic radiation modes are by definition the basis that orthogonalizes the radiation operator, meaning the radiation modes do not exhibit any coupling in radiation of sound.Acoustic radiation modes are coming up on their 30th anniversary in the literature, but still have not found wide use. This is largely due to the fact that most radiation modes must be calculated through the computationally intensive boundary element method or boundary integral equations. Analytical expressions for radiation modes, or for the radiation resistance matrix from which they are derived, are only available for a few geometries. This thesis meets this problem head on, to develop additional analytical expressions for radiation resistance matrices of cylindrically curved structures.Radiation modes are developed in the context of their use to calculate sound power. Experimental and computational sound power calculations are presented in order to validate the use of the modes developed here. In addition, the properties and trends of the developed modes are explored.

Travis Nathan Hoyt (Masters Thesis, August 2019, Advisor: Scott Sommerfeldt )

Abstract

Sound power measurements of acoustic sources are typically performed in anechoic or reverberation chambers using acoustic pressure according to international standards. The anechoic chamber creates a free-field environment where the sound power is estimated from the squared pressure integrated over some enveloping surface. The reverberation chamber produces diffuse-field conditions, where sound power is proportional to the spatially averaged squared pressure. In semi-reverberant environments, the direct and reverberant energies each contribute to the total measured field. If the kinetic and potential components of acoustic energy density are weighted appropriately, the spatial variation of the field can be significantly reduced compared to squared pressure. This generalized energy density allows an adaptation of the sound power formulation by Hopkins and Stryker to be used to make an efficient and accurate in situ sound power estimate of a noise source in a non-ideal acoustical environment. Since generalized energy density optimizes the spatial uniformity of the field, fewer measurement positions are needed compared to traditional standards. However, this method breaks down for sources that are large and extended in nature and considerably underestimates the sound power. This thesis explores the practical limits of this method related to the sound power underestimation. It also seeks to understand the special considerations necessary to achieve accurate, survey-grade sound power data of large, extended noise sources through a laboratory study of custom extended and compact sources. A modified method to accurately and efficiently measure the sound power of large, extended sources is proposed with results.

Jared Miller (Capstone, April 2019, Advisor: Scott Sommerfeldt )

Abstract

There has always been a significant population concerned with achieving the best possible environment for listening to music; but in our world of convenience, people don’t want to have to go to the symphony hall to enjoy such an experience. The research presented here uses active noise control to alter sound produced in a given environment, in order to match a more desirable listening environment. This is done by incorporating a measured acoustic impulse response, representing the desired listening environment, into a filtered-x algorithm. The algorithm adapts until the response in the actual listening space matches the response of the desired listening space. Creating a computational simulation allowed for rigorous testing of the algorithm. Once the algorithm was fine tuned through the simulated testing, it could be applied in a physical system. Initial physical testing shows promising results, so further investigations are recommended.

Michael Thomas Rose (Masters Thesis, June 2019, Advisor: Scott Sommerfeldt )

Abstract

[Abstract]

2017

Pegah Aslani (PhD Dissertation, August 2017, Advisor: Scott Sommerfeldt )

Abstract

Cylindrical shells are common structures that are often used in industry, such as pipes, ducts, aircraft fuselages, rockets, submarine pressure hulls, electric motors and generators. In many applications it is desired to attenuate the sound radiated from the vibrating structure. There are both active and passive methods to achieve this purpose. However, at low frequencies passive methods are less effective and often an excessive amount of material is needed to achieve acceptable results. There have been a number of works regarding active control methods for this type of structure. In most cases a considerable number of error sensors and secondary sources are needed. However, in practice it is much preferred to have the fewest number of error sensors and control forces possible. Most methods presented have shown considerable dependence on the error sensor location. The goal of this dissertation is to develop an active noise control method that is able to attenuate the radiated sound effectively at low frequencies using only a small number of error sensors and secondary sources, and with minimal dependence on error sensor location. The Weighted Sum of Spatial Gradients control metric has been developed both theoretically and experimentally for simply supported cylindrical shells. The method has proven to be robust with respect to error sensor location. In order to quantify the performance of the control method, the radiated sound power has been chosen. In order to calculate the radiated sound power theoretically, the radiation modes have been developed for cylindrical shells. Experimentally, the radiated sound power without and with control has been measured using the ISO 3741 standard. The results show comparable, or in some cases better, performance in comparison with other known methods. Some agreement has been observed between model and experimental results. However, there are some discrepancies due to the fact that the actual cylinder does not appear to behave as an ideal simply supported cylindrical shell.

Kelli Fredrickson Succo (Masters Thesis, December 2017, Advisor: Scott Sommerfeldt )

Abstract

The phase and amplitude gradient estimator (PAGE) method has proven successful in improving the accuracy of measured energy quantities over the p-p method, which has traditionally been used, in several applications. One advantage of the PAGE method is the use of phase unwrapping, which allows for increased measurement bandwidth above the spatial Nyquist frequency. However, phase unwrapping works best for broadband sources in free-field environments with high coherence. Narrowband sources often do not have coherent phase information over a sufficient bandwidth for a phase unwrapping algorithm to unwrap properly. In fact, phase unwrapping processing can cause significant error when there is no coherent signal near and above the spatial Nyquist frequency. However, for signals at any frequencies up to the spatial Nyquist frequency, the PAGE method provides correct intensity measurements regardless of the bandwidth of the signal. This is an improved bandwidth over the traditional method. For narrowband sources above the spatial Nyquist frequency, additional information is necessary for the PAGE method to provide accurate acoustic intensity. With sufficient bandwidth and a coherence of at least 0.1 at the spatial Nyquist frequency, a relatively narrowband source above the spatial Nyquist frequency can be unwrapped accurately. One way of using extra information, called the extrapolated PAGE method, uses the phase of a tone below the spatial Nyquist frequency and an assumption of a propagating field, and therefore linear phase, to extrapolate the phase above the spatial Nyquist frequency. Also, within certain angular and amplitude constraints, low-level broadband noise can be added to the field near a source emitting a narrowband signal above the spatial Nyquist frequency. The low-level additive broadband noise can then provide enough phase information for the phase to be correct at the frequencies of the narrowband signal. All of these methods have been shown to work in a free-field environment.

2016

Matthew Franklin Calton (Masters Thesis, August 2016, Advisor: Scott Sommerfeldt )

Abstract

[Abstract]

2014

Candice Marie Humpherys (Masters Thesis, June 2014, Advisor: Scott Sommerfeldt )

Abstract

In contrast to the structural modes, which describe the physical motion of vibrating structures, acoustic radiation modes describe the radiated sound power. Radiation modes are beneficial in active noise control because reducing an efficiently radiating radiation mode guarantees the reduction of radiated sound power. Much work has been done to calculate the radiation modes for simple geometries, where analytic solutions are available. In this work, isogeometric analysis (IGA) is used to provide a tool capable of analyzing the radiation modes of arbitrarily complex geometries. IGA offers increased accuracy and efficiency by using basis functions generated from Non-Uniform Rational B-Splines (NURBS) or T-Splines, which can represent geometries exactly. Results showing this increased accuracy and efficiency with IGA using T-Splines are shown for a sphere to validate the method, comparing with the exact analytical solution as well as results from a traditional boundary element method. A free cylindrical shell is also analyzed to show the usefulness of this method. Expected similarities, as well as expected differences, are observed between this free shell and a baffled cylindrical shell.

Daniel Ryan Marquez (Masters Thesis, March 2014, Advisor: Scott Sommerfeldt )

Abstract

Sound power measurements of acoustic sources are generally made in reverberation or anechoic chambers using acoustic pressure measurements as outlined in specific ISO or other standards. A reverberation chamber produces an approximate diffuse-field condition, wherein the sound power is determined from the spatially averaged squared pressure. An anechoic chamber produces an approximate free-field condition, wherein the sound power is estimated from squared pressure over an enveloping measurement surface. However, in many cases it is desirable to estimate sound power within nonideal semi-reverberant spaces. In these environments, both direct and reverberant energies may contribute significantly to the total acoustic field. This paper introduces two measurement methods that utilize a weighted combination of potential and kinetic energy densities, known as generalized acoustic energy density, to estimate sound power in nonideal semi-reverberant rooms. The first method employs a generalized sound power formulation, which is an adaptation to an equation developed in 1948 for semi-reverberant spaces. The second, called the two-point in situ method, is a technique based on the generalized sound power formulation for quick and accurate in situ sound power estimates. Since the generalized acoustic energy density is more spatially uniform than the squared acoustic pressure in an enclosed field, these methods have the advantage of achieving the same accuracy in sound power determination with fewer measurement positions. This thesis explores the possibility of using these new methods in place of methods outlined in current ISO standards by describing analytical, numerical, and experimental results.

2013

Zachary Jensen (Senior Thesis, April 2013, Advisor: Scott Sommerfeldt )

Abstract

Acoustic enclosures are commonly used to attenuate the noise radiated from a sound source. A model to accurately predict the insertion loss of an enclosure can quickly become complicated for complex configurations and have uncertainties over different frequency ranges. In many such situations, a reliable, simple method of predicting insertion loss with reasonable accuracy would be valuable if a quick estimation is needed for a certain enclosure. An insertion loss model was tested using a Design of Experiments (DOE) approach, which incorporated a range of apertures, absorptive treatment, and obstructions, with varying surface areas and positions in order to span a large design space. This paper discusses the insertion loss measurements of the DOE test configurations and compares the measurements with an attempted fit of all 36 configurations. Nearly 80% of the configurations fell below an error of 2.5 dB, and 100% fell below 4.5 dB, which was deemed to provide a reasonable rough estimate of insertion loss.

Brent Reichman (Senior Thesis, April 2013, Advisor: Scott Sommerfeldt )

Abstract

Active noise control (ANC) uses a control signal to effectively cancel out unwanted sound. Applying ANC to snoring presents an interesting challenge because of its unpredictable nature and the close distance between the source and the desired region of cancellation. This experiment focuses on two factors: How much attenuation can be achieved in a standard bed using different microphone and speaker setups and how large is the "zone of silence" that is created.

2012

John Boyle (Capstone, June 2012, Advisor: Scott Sommerfeldt )

Abstract

This paper presents an experimental verification of the error sensor placement theory developed by Esplin (2012) for control of the blade passage frequency (BPF) tone in the exhaust duct of a notebook computer centrifugal fan. As the theory modeled the fan and duct as a baffled, closed-open, rectangular duct, such a duct was constructed. Small receivers were placed in the side of the duct at locations similar to those of the primary noise source of the fan and the secondary (control) source used in the model. Maps of the pressure difference in the plane of the baffle between the pressure field of the primary source and the field corresponding to the predicted minimum sound power condition were used to guide error sensor placement. Radiated sound power reduction of tones in the baffled, closed-open duct was measured and showed good agreement with predicted values. The model predictions were applied to the real fan and exhaust duct, and sound power reduction was measured. The observed trend of power reduction over frequency follows prediction, and suggests the simplified model of the fan assembly is an effective approximation for ANC purposes.

Cole Victor Duke (Masters Thesis, July 2012, Advisor: Scott Sommerfeldt )

Abstract

Active noise control (ANC) has been implemented using analog filters to reduce broadband noise from a small axial cooling fan. Previous work successfully attenuated narrow-band, tonal portions of the noise using a digital controller. The practical performance limits of this system were reached and it was desirable to attenuate the noise further. Additional research, therefore, sought to attenuate broadband noise from the fan using a digital controller, but performance was limited by the group delay inherent in the digital signal processor (DSP). Current research attempts to further attenuate broadband noise and improve performance of the system by combining the tonal controller with an analog feedback controller. An analog controller is implemented in parallel with the digital controller without degrading the performance of either individual controller. Broadband noise is attenuated in a certain frequency region, but at the expense of increasing noise in adjacent frequency regions. Results show that a single-input single-output (SISO) controller is preferable to a multiple-input multiple-output (MIMO) controller for this system.

John J Esplin (Masters Thesis, June 2012, Advisor: Scott Sommerfeldt )

Abstract

Noise from information technology (IT) equipment is a significant problem in today’s modern society. Active Noise Control (ANC) has shown promise in reducing the effect of IT fan noise on users. Though ANC has been applied to axial fans (such as those found in desktop computers), it has not been applied to centrifugal fans, such as those found in laptop computers. This work applies an ANC method to a centrifugal fan mounted in a mock laptop enclosure. This method is applied in four steps. First, secondary sources are placed in the vicinity of the fan. Second, an accurate model of the radiation from the fan and secondary sources is constructed. Third, the total power radiated from this system is minimized. This creates nodal lines in the vicinity of the fan. Fourth, ANC error sensors are placed on the nodal lines predicted by the model. This creates these nodal lines experimentally, thus creating the minimum power condition. The noise from the exhaust and inlets of the fan will first be controlled individually. Then the method will be applied to the combined system. Global sound power radiation will be measured in all cases.

2010

Buye Xu (PhD Dissertation, September 2010, Advisor: Scott Sommerfeldt )

Abstract

The properties of acoustic kinetic energy density and total energy density of sound fields in lightly damped enclosures have been explored thoroughly in the literature. Their increased spatial uniformity makes them more favorable measurement quantities for various applications than acoustic potential energy density (or squared pressure), which is most often used. In this dissertation, a new acoustic energy quantity, the generalized acoustic energy density (GED), will be introduced. It is defined by introducing weighting factors, α and 1 − α, in the formulation of total acoustic energy density. With the additional degree of freedom, the GED can conform to the traditional acoustic energy density quantities, or be optimized for different applications. The properties and applications of the GED are explored in this dissertation. For enclosed sound fields, it was found that GED with α = 1/4 is spatially more uniform than the acoustic potential energy density, acoustic kinetic energy density, and the total acoustic energy density, which makes it a more favorable measurement quantity than those traditional acoustic energy density quantities for many indoor measurement applications. For some other applications, such as active noise control in diffuse field, different values of α may be considered superior. The numerical verifications in this research are mainly based on a hybrid modal expansion developed for this work, which combines the free field Green’s function and a modal expansion. The enclosed sound field is separated into the direct field and reverberant field, which have been treated together in traditional modal analysis. Studies on a point source in rectangular enclosures show that the hybrid modal expansion converges notably faster than the traditional modal expansions, especially in the region near the source, and introduces much smaller errors with a limited number of modes. The hybrid modal expansion can be easily applied to complex sound sources if the free field responses of the sources are known. Damped boundaries are also considered in this dissertation, and a set of modified modal functions is introduced, which is shown to be suitable for many damped boundary conditions.

2008

J. Isaac Fjeldsted (Capstone, August 2008, Advisor: Scott Sommerfeldt )

Abstract

In recent years there has been a greater demand in industry for noise control on large construction vehicles. The Brigham Young University Acoustics Group has been contracted to reexamine insertion loss equations that are currently being used to model the sound field from engine enclosures of these types of vehicles. It has been hypothesized that inaccuracies in the model are partly due to the fact that diffraction is not considered. A numerical model of simple diffraction by an aperture has been created and needs to be tested for accuracy. The purpose of this research is to gather experimental data in order to assess how well this model predicts actual diffracted sound fields from an aperture. Error analysis between experimental and predicted data shows that the numerical model being tested is accurate within a tolerance of ± 4 dB. Research indicates that this model of diffraction may be used in further research to understand insertion loss of engine enclosures.

Stephan P Lovstedt (Masters Thesis, March 2008, Advisor: Scott Sommerfeldt )

Abstract

The Filtered-X Least-Mean-Square (FXLMS) algorithm is widely used in active noise control due to its robustness, simplicity, and ability to be implemented in real time. In a feedforward implementation of the FXLMS algorithm, a reference signal that is highly correlated with the noise to be controlled is filtered with an estimate of the transfer function of the secondary path. The convergence characteristics of the FXLMS algorithm have been well studied. A convergence parameter is used to optimize the convergence of the algorithm. However, the optimal value for the convergence parameter is frequency dependent. Thus for noise containing multiple tones at different frequencies the convergence parameter can only be optimized for one of those tones. Other tones will have slower convergence rates and in general less attenuation than they would have if they were treated singly and parameters could be optimized for those frequencies separately. A method is developed to modify the magnitude response of the secondary path estimate while maintaining the original phase response, which equalizes the convergence characteristics over multiple frequencies, giving more uniform convergence and attenuation for all tones being controlled. Stability of the algorithm is not compromised. The modification to the FXLMS algorithm is relatively simple to implement and has been shown to increase overall attenuation of a signal containing multiple tones by an additional 6-9 dB.

2007

Cole Duke (Capstone, April 2007, Advisor: Kent Gee, Scott Sommerfeldt )

Abstract

In the active control of tonal noise from cooling fans, one factor that can limit the achievable attenuation is fluctuation of the blade passage frequency in time. Large fluctuations in a short time can hinder the algorithm from converging to the optimal solution. Some fans have steadier speeds than others, which can be due to unsteady driving mechanisms or the physical structure of the fan. Environmental effects such as back pressure and unsteady blade loading can also cause the fan speed to fluctuate. The shifting in the blade passage frequency will be measured using a zero-crossing technique to track the frequency of each cycle. Blade passage frequency fluctuations will be presented for various driving mechanisms and environmental conditions. Techniques to minimize frequency shifting will also be discussed.

Connor Raymond Duke (Masters Thesis, December 2007, Advisor: Scott Sommerfeldt )

Abstract

"Previous work has shown that active noise control (ANC) can be applied to axial cooling fans. Optimization of the control source and error sensor placement is desired to maximize the attenuation using ANC. A genetic algorithm was developed to find the optimal placement of control sources for a given primary source. The optimal configuration of control sources around a single primary source was shown to be a linear arrangement of the sources. This holds true for both two-dimensional as well as threedimensional configurations. The higher-order radiation of the linear arrangement has also been verified experimentally, but the improvement in the experimental apparatus was not as dramatic as the theoretical model. Multiple flow visualization techniques have been used to find optimal near field error sensor locations. When there is little obstruction to the flow field of the fan, minimal airflow is found along the near field null that is created by minimizing the sound power of the system. Surface mounting of the error sensors can lead to a small increase in the signal-to-noise ratio of the error sensors if

2006

Matthew J Green (Masters Thesis, August 2006, Advisor: Scott Sommerfeldt )

Abstract

Feedback active noise control (ANC) has been applied as a means of attenuating broadband noise from a small axial cooling fan. Such fans are used to maintain thermal stability inside of computers, projectors, and other office equipment and home appliances. The type of low-level noise radiated from axial cooling fans has been classified as harmful to productivity and human well being. Previous research has successfully implemented feed-forward ANC, targeting specific narrow-band fan noise content related to the blade passage frequency (BPF) of the fan. The reference signal required for a feed-forward algorithm limits its ability to attenuate much of the noise content; however, it is also desirable to reduce broadband fan noise. Feedback control is a logical alternative in the absence of a valid reference signal.

Brian B Monson (Masters Thesis, July 2006, Advisor: Scott Sommerfeldt )

Abstract

Previous work has shown that active noise control is a feasible solution to attenuate tonal noise radiated by small axial cooling fans, such as those found in desktop computers. One such control system reduced noise levels of a baffled 80-mm fan in the free field with four small loudspeakers surrounding the fan. Due to industry specified spatial constraints, a smaller fan and speaker configuration was desirable. The smaller configuration maintains similar control performance, further facilitating practical implementation of the control system. The smaller control system employs a smaller fan running at a higher speed. Different loudspeaker configurations for control exist and have been tested. A configuration consisting of four control sources spaced symmetrically around and coplanar to the fan exhibits global control of the tonal component of the fan noise. A configuration with three symmetrically spaced sources is shown to perform similarly, agreeing with theoretical prediction. An analysis of the control system in a non-ideal reflective environment is also discussed.

2005

Michael Dickerson (Capstone, December 2005, Advisor: Timothy Leishman, Scott Sommerfeldt )

Abstract

A single-string instrument (monochord) was carefully designed and constructed to test a new prototype of a two-dimensional magnetic pickup. The objective of the research was to design the new transducer which would depict a more meaningful signal analogous to the two-dimensional vibration of the string. The monochord and 2D pickup were designed with much specification and detail. However, the testing of the output power spectra and frequency response functions to validate performance proved insufficient. Although the results were inconclusive, extensive progress was made toward determining the correct methods needed to test the true performance of the pickup and can be continued.

2004

Joel Andrus (Capstone, April 2004, Advisor: Scott Sommerfeldt )

Abstract

Sound Power is a widely used quantity used to describe how much sound an acoustic source is radiating. It is more commonly used than Sound Pressure Level because it is independent of the measurement location. Sound Power, which is measured in watts (W), is often expressed on a logarithmic scale. This gives the Sound Power Level, which has the units of decibels (dB). Sound Power Level can be determined by finding the average Sound Pressure Level at several points in the far field surrounding the source, which can be found from several different ways. Since the square of the pressure in the far field is proportional to the intensity, it can be integrated to get sound power. Sound Power can also be determined by scanning over a surface with an intensity probe, and then integrating the intensity over the area scanned. Another possible way to measure sound power is with an energy-density probe. This is the method being investigated in this paper.

Benjamin Mahonri Faber (Masters Thesis, March 2004, Advisor: Scott Sommerfeldt )

Abstract

An active noise control (ANC) system has been applied to the problem of attenuating low-frequency tonal noise inside small enclosures. The intended target application of the system was the reduction of the engine firing frequency inside heavy equipment cabins. The ANC system was based on a version of the filtered-x LMS adaptive algorithm, modified for the minimization of acoustic energy density (ED), rather than the more traditional minimization of squared acoustic pressure (SP). Three loudspeakers produced control signals within a mock cabin composed of a steel frame with plywood sides and a Plexiglas® front. An energy density sensor, capable of measuring acoustic pressure as well as acoustic particle velocity, provided the error signal to the control system. The ANC system operated on a single reference signal, which, for experiments involving recorded tractor engine noise, was derived from the engine’s tachometer signal. For the low frequencies at which engine firing occurs, experiments showed that ANC systems minimizing ED and SP both provided significant attenuation of the tonal noise near the operator’s head and globally throughout the small cabin. The tendency was for ED control to provide a more spatially uniform amount of reduction than SP control, especially at the higher frequencies investigated (up to 200 Hz). In dynamic measurement conditions, with a reference signal swept in frequency, the ED control often provided superior results, struggling less at frequencies for which the error sensor was near nodal regions for acoustic pressure. A single control channel often yielded performance comparable to that of two control channels, and sometimes produced superior results in dynamic tests. Tonal attenuation achieved by the ANC system was generally in excess of 20 dB and reduction in equivalent sound level for dynamic tonal noise often exceeded 4 dB at the error sensor. It was shown that temperature changes likely to be encountered in practice have little effect on the initial delay through the secondary control path, and are therefore unlikely to significantly impact ANC system stability in the event that a fixed set of system identification filter coefficients are employed.

Lance Lester Locey (Masters Thesis, December 2004, Advisor: Scott Sommerfeldt )

Abstract

"Traditional methods for the investigation of sound fields generally rely on a microphone to convert sound pressure into an electrical signal which can be recorded, displayed, and so forth. The squared sound pressure is directly related to potential energy density. Consequently, the measurement of sound pressure alone does not inherently provide insight into the total energy density of the sound field. Specifically, no information about the kinetic energy density of the sound field is available from this measurement alone. However, it is possible to use two microphones to estimate particle velocity. The squared particle velocity magnitude is directly related to kinetic energy density. The two energy quantities combine to yield total acoustic energy density. The purpose of this work is to investigate and compare three probes designed to measure acoustic energy density. It is also to determine which probe may be most practically implemented in real-world applications. All three designs are based on a rigid spherical housing and are referred to as follows: the six microphone probe, the

2003

Gordon Dix (Capstone, April 2003, Advisor: Scott Sommerfeldt )

Abstract

Intensity is a valuable tool in acoustic measurements for source identification and sound power levels, but most commercially available intensity probes on the market today consist of only one pair of phase-matched microphones to measure one axis at a time. A fully automated intensity measurement system was needed in the Acoustic Testing Lab at NASA Glenn to reduce the time of measurements and to increase the ability to identify noise problems at specific frequencies. A two-dimensional intensity probe was designed and machined at Brigham Young University and a gantry system was developed to guide the probe throughout the anechoic chamber. The system is capable of scanning planes as large as 10 feet in the vertical direction, and more in the horizontal directions. Multiple spacers were included with the probe to increase the frequency range available for each test. Labview based software was also written to handle the data acquisition and motion control of the system. The data was displayed using two methods. A color plot maps the magnitude over the measurement grid, while a modified quiver plot shows the direction and magnitude of the data. System parameters and capabilities will be presented as well as sample test data.

Angela Tygerson (Senior Thesis, April 2003, Advisor: Scott Sommerfeldt )

Abstract

2002

Kent L Gee (Masters Thesis, August 2002, Advisor: Scott Sommerfeldt )

Abstract

A multi-channel active control system has been applied to the reduction of free-field tonal noise from a small axial cooling fan, typical of those used in office equipment and other technology. The experimental apparatus consists of an aluminum enclosure which houses the fan, an infrared detector-emitter pair which serves as a reference sensor, loudspeakers, microphones, and appropriate filters and amplifiers. The control signals are generated using a single reference, multiple output filtered-x algorithms. Potential near field error microphone locations have been investigated by modeling the research fan and loudspeakers as point sources to obtain a mathematical expression for radiated power. The minimization of this power has yielded likely microphone locations in the extreme near field and has also guided the number and location of control sources. Experiments with various control configurations have shown multiple control channels are required for significant global attenuation of higher harmonics of the research fan’s blade passage frequency (BPF). In addition, there are predictable near field locations which consistently lead to significant reductions in the global mean-square pressure (MSP) for the first four harmonics of the BPF. For example, a four channel configuration results in MSP reductions of 9-18 dB for each of the four harmonics.

Laralee Gordon Ireland (PhD Dissertation, December 2002, Advisor: Scott Sommerfeldt )

Abstract

A numeric model was developed to investigate the possibility of implementing active control (ANC) to minimize noise radiation form high-bypass turbofan engines. Previous experimental work on the NASA Glenn Research Center active noise control fan (ANCF) was encouraging, but the question remained whether the modal approach investigated could be effective on real engines. The engine model developed for this research project uses an indirect b9oundary element method, implemented with Sysnoise, and a multi-mode Newton’s algorithm, implemented with MATLABTM, to simulate the active control. Noise from the inlet was targeted. Both the experimental and numerical results based on the NASA ANCF simplified cylindrical engine geometry indicate overall reductions in the m=2 component of the noise. Reductions obtained at the numerical sensor rings range from 17 dB to 63 dB and at a plane in the duct inlet, -8dB to 33 dB. Tings mounted on the inlet duct are unable to accurately predict the total reduction of the inlet field, but the controller is still able to effectively reduce the total acoustic field. Generally, one senor ring and one actuator ring per propagating mode were necessary to control the inlet field. At frequencies close to the cut-off frequency of a mode, an additional sensor and actuator ring were needed to adequately control the inlet field due to the evanescent mode. A more realistic, but still axisymmetric, engine geometry based on the GE CF6-80C engine was developed and the same algorithm used. Reductions obtained at the sensor rings range from 4 dB to 56 dB and at the duct inlet plane, from 12 dB to 26 dB. The overall far field noise radiation form the engine remained unchanged (0.4dB) or decreased slightly (3.6 dB). The inlet noise was controlled at all frequencies but the noise from the exhaust was increased. The effect of inlet control on the exhaust radiation suggests the need for a controller that targets both the inlet and exhaust noise simultaneously. The results of this simulation indicate that modal ANC approach should still be effective for controlling turbofan noise in a more realistic engine geometry.

Stephen W. Lee (Honors Thesis, January 2002, Advisor: Scott Sommerfeldt )

Abstract

Heather McKnight Smith (Senior Thesis, April 2002, Advisor: Scott Sommerfeldt )

Abstract

2000

Vivian Francis Dias (Masters Thesis, December 2000, Advisor: Scott Sommerfeldt )

Abstract

A conventional active vibration control (AVC) system in its simplest form consists of three main components: a sensor (to detect the vibration), a controller (to manipulate the data and provide a control signal), and an actuator (to generate a secondary vibrational response to destructively interfere with the primary response at the sensor location). In typical AVC systems, the control algorithm usually minimizes the signal obtained from the error sensor at its location, in the least mean square sense. In some cases, however, it may be difficult to permanently locate the error sensor(s) at the location(s) where the optimum attenuation is desired. This research aims to address this problem for structures excited with a single frequency excitation using the approach of ‘virtual sensors’. Virtual sensors use information from physical sensor(s) at other locations on the structure and transfer functions to estimate the vibration response at the virtual sensor location. The research indicates that it is possible to achieve attenuation from 20dB - 38 dB depending on the desired location on the structure and the excitation frequency. These results are within 6-8 dB of the reduction achieved using the conventional error sensor technique. A sensitivity study done on the transfer functions used by the ‘virtual sensor’ technique showed that the technique generally has rather low sensitivity to the phase of the transfer functions and somewhat higher sensitivity to the magnitude of the transfer functions. Reasonably good results (i.e. reduction in vibration levels) were achieved in most cases even when the transfer functions used were 25-30% off the measured values. The concept of virtual sensors could be sued effectively for online maintenance of vibrating structures and enhancing the life of critical parts. Assuming that the location of the structural damage can be identified at its inception, one could apply active vibration control on the structure as a means of reducing stresses in the damage zone, thus ensuring that the damage does not grow rapidly with time.

Guillermo Aurelio Herrera (Masters Thesis, December 2000, Advisor: Scott Sommerfeldt )

Abstract

There is currently considerable interest in developing models that more accurately predict the response of systems with small loudspeakers, such as those in cellular telephones. There has also been a trend towards developing hands-free communication with cell-phones, thus requiring a relatively high acoustic output from a small source. This work is based on the development of a model for a cell phone to help address such research issues. It has been shown that boundary layer effects can cause significant frequency shifts in the resonance peaks associated with systems having elements with dimensions on the order of the boundary layer thickness. This phenomenon has been observed in the presence of ports with diameters of 1 mm or less. These effects, and the higher modes of the loudspeaker diaphragm, were incorporated in two cell phone earpieces and hands-free systems, obtaining good predictions of the frequency response within 2dB and 5dB, respectively, of the measured response.

1999

Dong Lin (PhD Dissertation, August 1999, Advisor: Scott Sommerfeldt )

Abstract

"Two methods for estimating the total acoustic power radiated by an infinitely baffled beam with various boundary conditions are studied numerically. One method is based no distributed strain sensors, while the second method is based on distributed displacement sensors. The objective is to use a minimal number of fixed sensors to obtain the acoustic power radiated form a beam under broadband excitation. The radiation mode concept is adopted in both methods. Previous research with radiation modes has generally focused on single frequency excitation at relatively low frequencies (i.e. kl is typically less than 2). In the first method investigated here, each senor is designed to measure a single radiation mode at a given frequency, and a linear interpolation scheme is used to estimate the broadband excitation. It is demonstrated that if the first M radiation modes radiate efficiently, then only 2M sensors are needed for boundary conditions that correspond to zero displacement. Computer simulation shows that for a one meter long beam, and with only the first four radiation modes considered, the estimated power is within 3dB of the exact power for single frequency excitation up to an excitation frequency of 650 Hz (kl = 12), and most of the time within 3dB up to 1.2kHz (kl = 22).

1998

Karl Kowallis (Honors Thesis, January 1998, Advisor: Scott Sommerfeldt )

Abstract

Timothy O Samuels (Masters Thesis, August 1998, Advisor: Scott Sommerfeldt )

Abstract

An attempt has been made to modify a standard active noise control algorithm in order to take into account the unique response of a human auditory system. It has been shown in the past that decreasing the sound pressure level at a location does not guarantee a similar decrease in the perceived loudness at that location. Typically, noise cancelation is based on the “error signal” from a mechanical device such as a microphone, whose response is nominally flat across the frequency response range of the human ear. However, if the response if the ear can be approximated by digitally filtering the error signal before it reaches the adaptive controller, one can, in effect, minimize the more subjective loudness level, as opposed to the sound pressure level. The work reported here entails simulating noise control based upon minimizing perceived loudness for a collection of input noise signals. A comparison of the loudness of the resulting error signal is made to the loudness of that resulting from standard sound pressure level minimization. It has been found that the effectiveness of this technique is largely dependent upon the nature of the input noise signal. Furthermore, this technique id worth considering for use with applications of noise control where the uncontrolled noise more prominently constitutes low range audio frequencies (approx. 30 Hz – 100 Hz) than medium range audio frequencies (approx.. 300 Hz – 600 Hz).

1995

Brian L Scott (Masters Thesis, January 1995, Advisor: Scott Sommerfeldt )

Abstract