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Department Library

2020

Dallin Haslam (Capstone, August 2020, Advisor: Timothy Leishman )

Abstract

Guitar amplifiers are specialized loudspeakers with complex configurations. Considerations of where a microphone is placed relative to an amplifier and where amplifiers are placed in a live venue or recording studio are affected by directivity of each individual amplifier. This paper presents measured transverse-plane polar, and interpolated spherical directivities of four guitar amplifiers in an anechoic space. Using this data one may observe how differences in amplifier configurations affect their directivity characteristics.

2019

Sam Bellows (Senior Thesis, April 2019, Advisor: Timothy Leishman )

Abstract

Directivity measurements of human speech reveal important characteristics of sound radiation and are useful in a variety of applications. Previous work in speech directivity has used different approaches; for this work, the directivity factors and indices of speech were measured using both a single and multiple-capture scanning system with human subjects. Analysis in the spherical harmonic domain helped to show important relationships between the different techniques as well as to simplify relations between measured frequency-response functions and the corresponding directivity factor and index functions. The high-resolution results show that while the directivity of human speech is omnidirectional at low frequencies, it becomes more directional at higher frequencies. Furthermore, diffraction lobes play a significant role in the directivity of human speech.

2018

Jennifer Kay Whiting (Masters Thesis, April 2018, Advisor: Timothy Leishman )

Abstract

This thesis describes the development of the real-time convolution system (RTCS) for a little-studied talker/listener in virtual acoustic environments. We include descriptions of the high-resolution directivity measurements of human speech, the RTCS system components, the measurement and characterization of oral-binaural room impulse responses (OBRIRs) for a variety of acoustic environments, and the compensation filter necessary for its validity. In addition to incorporating the high-resolution directivity measurements, this RTCS improved on that developed by Cabrera et al. [1] through the derivation and inclusion of the compensation filter. Objective measures in the time- and frequency-domains, as well as subjective measures, were developed to asses the validity of the RTCS. The utility of the RTCS is demonstrated in the study on vocal effort, and the results of an initial investigation into the vocal effort data are presented.

2017

Claire Pincock (Capstone, April 2017, Advisor: Timothy Leishman )

Abstract

High-resolution speech directivity data has been measured in the BYU anechoic chamber. The data was taken at 2,522 discrete points in a spherical measurement surface. While speech directivity data have been taken in the past, the current increased-resolution data are more useful in speech and architectural acoustic studies. This capstone report presents the results of the measurements in the forms of balloon plots and polar plots with interesting frequency-dependent trends. The directivity is nearly omnidirectional at lower frequencies such as 200 Hz. Higher levels below the mouth axis are present at mid-range frequencies such as 800 Hz. Higher radiation levels above the head are present at higher frequencies such as 1,600 Hz. In order to resample the balloon plots or begin the process of modeling directivity patterns at other radii, a spherical harmonic expansion method was developed so the discrete measurement data could be described in terms of continuous functions.

2016

K Joshua Bodon (Masters Thesis, April 2016, Advisor: Timothy Leishman )

Abstract

A high-resolution directivity measurement system at Brigham Young University has been renovated and upgraded. Acoustical treatments have been installed on the microphone array, professional-grade audio hardware and cabling have been utilized, and user-friendly MATLAB processing and plotting codes have been developed. The directivities of 16 played musical instruments and several loudspeakers have been measured by the system, processed, and plotted. Using loudspeakers as simulated musicians, a comprehensive analysis was completed to validate the system and understand its error bounds. A comparison and evaluation of repeated-capture to single-capture spherical systems was made to demonstrate the high level of detail provided by the 5 degree resolution system. Analysis is undertaken to determine how nonanechoic effects in anechoic chambers influence results. An overview of directivity measurement systems from the literature is provided as well as a dedicated discussion of the directivity measurement system at Brigham Young University.

Michael Denison (Capstone, April 2016, Advisor: Timothy Leishman )

Abstract

Anechoic chambers are typically qualified by comparing sound pressures at several radial distances from a sound source and verifying that they follow the spherical spreading law within specified tolerances. While this technique is useful, it may not sufficiently characterize free-field variations at fixed radial distances and numerous angular positions, as are commonly used for directivity, sound power, and other important acoustical measurements. This paper discusses a technique to detect angular field deviations in anechoic chambers. It incorporates a loudspeaker in an altazimuth mount, an adjustable-radius boom arm, and a precision microphone. The boom arm and microphone remain in line with the loudspeaker driver axis at a fixed radius while the system rotates to specified azimuthal angle increments. In an ideal free-field environment, the frequency-response function from the loudspeaker input signal to the microphone output signal should remain consistent—regardless of the system orientation. However, in typical anechoic chambers, they vary. Standard deviation calculations over many angles reveal frequency-dependent departures from the ideal, especially for narrow-band data. The results show the impact of these discrepancies for multiple-angle measurements and how they change with radial distance from the source.

Travis Hoyt (Capstone, August 2016, Advisor: Timothy Leishman )

Abstract

The volume velocity of an acoustic source is its rate of sound-induced flow of the medium (typically air) through a specified surface. The measurement of this acoustic flux is used by some industry researchers to evaluate the acoustics of their products. Current experimental methods used to determine the volume velocity of loudspeakers involve tedious and costly procedures such as scanning laser Doppler vibrometry in an anechoic environment that are impractical in most situations. BYU researchers have developed a theoretical method that has been experimentally verified to describe the in situ volume velocity of a single-driver loudspeaker using only electrical data if certain premeasured parameters are known. This work investigates the application of the single-driver theory to multiple-driver loudspeaker arrays such as 12-driver dodecahedrons. The limits of the method were assessed and detailed. The theory, experimental methods, calculations, and results are discussed. Acceptable agreement has been found in multi-driver loudspeakers between the measured values and those predicted by this method within the bandwidths of interest. Experiments demonstrate that the volume velocities of these loudspeakers are fairly insensitive to typical changes in acoustic environments. A single 12-source dodecahedron was prototyped and tested as part of this work.

Zachary R Jensen (Masters Thesis, April 2016, Advisor: Timothy Leishman )

Abstract

The two-point in situ method is a technique for measuring the room constant of a semi-reverberant room and the sound power of a source in that room simultaneously using two measurement positions. Using a reference directivity source, where the directivity factor along any given axis of the source has been measured, one is able to use the Hopkins-Stryker equation to measure both the room constant and the sound power level of another source rather simply. Using both numerical and experimental data, it was found that by using generalized energy density (GED) as a measurement quantity, the results were more accurate than those using squared pressure. The results also improved when one measurement position was near the source and the other measurement position was far from the source. This resulted in strong contribution of both the direct and reverberant fields in each of the measurement positions. Another improvement to the two-point method was the use of a local, spatial average around the measurement position. The assumptions in the Hopkins-Stryker equation rely on this average and it was found that a small local spatial average improved the measurements. However, this improvement was greater for squared pressure than for GED. Several source sound power levels and room constants were measured to show that these measurements are improved by using the suggested techniques.

Michael Rollins (Capstone, August 2016, Advisor: Timothy Leishman )

Abstract

Schoolteachers have a particularly high rate of voice-problem symptoms. Room acoustics could be a significant reason for this prevalence, but more needs to be known about the effects room acoustics on vocal effort. With increased understanding, rooms could be designed to mitigate unhealthy vocal effort, and by extension, voice problems. The present study attempts to measure the influence of room acoustic parameters on vocal changes by comparing the vocal effort of typical talkers in several distinct acoustic environments. Thirty-two participants were recorded completing a battery of speech tasks in eight widely ranging acoustic conditions. Key vocal parameters were derived from associated recordings and the statistical significance of the influence of the room acoustic parameters on each of the vocal parameters was determined using standard one-way ANOVA tests. It was found that changes in EDT, C50, and %ALcons had highly correlated effects on several vocal parameters, notably smoothed cepstral peak prominence, acoustic vocal quality index (AVQI), and pitch strength. As the EDT increased and C50 and %ALcons decreased, these and other vocal parameters tended toward more dysphonic phonation. There were also gender differences in several vocal parameters, including AVQI, pitch strength, and other vocal effort-related parameters, with females tending to exert more vocal effort. These findings begin to objectify the effect of room acoustics on vocal accommodations and provide grounds for developing future talker-oriented room acoustical standards.

2015

Mark Berardi (Senior Thesis, December 2015, Advisor: Timothy Leishman )

Abstract

Several acoustical measures have previously been used to evaluate vocal effort. They are useful in evaluating occupational risk for teachers and other occupational voice users. Acoustical measures are also used in clinical speech-language pathology as inexpensive and noninvasive ways of evaluating pathology severity and tracking therapy progress. One such measure of vocal effort is the analysis of vocal onsets. In this paper, a single acoustic m based on relative fundamental frequencies of glottal pulses following voiceless consonants (the onset coefficient) is used to evaluate vocal effort in response to changes in background noise and reverberation time within speaking environments. Analysis shows that females and males have similar vocal effort levels in the most typical acoustical conditions. However, females respond to louder background noise and longer reverberation times with more changes to their vocal onsets than do males.

Nathan Eyring (Senior Thesis, July 2015, Advisor: Timothy Leishman )

Abstract

The main purpose of this thesis is to provide an introduction into the methodology of performing subjective analysis of acoustic sources. The author has found that it is not uncommon for those with a technical background to overlook the unique challenges associated with subjective assessments. The thesis describes the ethical procedures of working with human participants and gives aid in creating an IRB proposal. It then provides a brief introduction into how to perform statistical analysis. Finally, it discusses how to develop, evaluate, and implement a subjective evaluation. The appendixes provide examples of the author’s work in this area. It includes a listener training program that was developed and implemented at BYU, along with an analysis of the results. It also includes a subjective evaluation performed to evaluate the impact of new directivity data in room acoustic modeling software.

2014

Matthew Calton (Capstone, April 2014, Advisor: Timothy Leishman )

Abstract

This report explains an in-depth study of the acoustical properties of a local high school auditorium for a new loudspeaker system. The auditorium was modeled using computer simulation software to obtain estimates of the existing conditions. When the initial model of the auditorium was complete, detailed measurements of the hall were taken and compared to the model. The model was then adjusted until it was brought into agreement with the measured values. Once the model adequately described the actual space, it was used to simulate different loudspeaker systems until the optimal configuration was found.

Joseph Morris (Capstone, July 2014, Advisor: Timothy Leishman )

Abstract

The goal of this project was to master the enhanced acoustic simulator for engineers (EASE) and use it to effectively model the acoustics of the BYU's Royden G. Derrick Planetarium. Understanding and visualizing the complicated sound field in the room are important steps to creating a better design and implementation of a proposed surround sound system. Without the perspective the modeling offers, the quality of the sound system will suffer. To model this unique room, various scaling and perspective measures related to the building plans and the software program had to be taken into account. These required intuition and familiarity with architectural design tools and concepts. The design of the planetarium makes modeling difficult because of inherent limitations in EASE. However, circumventing the limitations was possible, meaning a clean model could be created to assess the acoustical behaviors in the dome and audience areas.

Jeshua Mortensen (Senior Thesis, April 2014, Advisor: Timothy Leishman )

Abstract

This paper presents a rapid modeling tool to acquire the directivities of the Platonic-solid loudspeakers. Platonic-loudspeakers have been widely used in architectural acoustical measurements as omnidirectional sources of sound; however, despite their highly symmetrical properties they become multidirectional as the frequency increases past their cutoff frequencies at about one kHz. When used to represent an omnidirectional point source in computational models the multidirectional quality is ignored, and a point source is used instead. Therefore, the purpose of this research is threefold: 1 give a better model than a point source, one that includes the multidirectional behavior that can be computed with rapidity; 2 to determine what effects the directivities, some contributing factors to being more omnidirectional; and 3 determine, if there is, a particular geometry of the Platonic solids that gives the most omnidirectional behavior. The dodecahedron is the most widely used of the five Platonic-solid geometries, yet there are other solids that for certain bandwidths outdo the dodecahedron as to their omnidirectional behavior, such as the tetrahedron.

Jenny Whiting (Senior Thesis, April 2014, Advisor: Timothy Leishman )

Abstract

The purpose of this research was to create a real-time convolution system for use in auralizations and to study subject behavior. To do this, models of real acoustical spaces were first created using the EASE software package. Impulse responses with a single collocated speaker and listener designed to represent a human subject were then generated from those models. These impulse responses were convolved in real time with voice signal from a live subject in the anechoic chamber. The resultant auralization was sent to the subject’s ears via offear headphones in real time, so as to create the illusion of physically being in an acoustical space different from the anechoic chamber. The primary result of this research was the successful implementation of the convolution system with models from EASE in the anechoic chamber at BYU. Many future studies will involve this convolution system, including studies of human perception of their own speech, and musican/performer perception of their own sound in various acoustic environments.

2011

Daniel Marquez (Capstone, August 2011, Advisor: Timothy Leishman )

Abstract

Analogous circuits are often used to model the properties of moving-coil loudspeakers. Recently, several new models have been proposed to better estimate their blocked electrical impedances. This research compares three modeled estimates to values based on two indirect measurements and one direct measurement. The former are determined through a complex curve-fitting routine and measured input impedance values, while the latter result from experimental techniques incorporating a vacuum system, a scanning laser Doppler vibrometer (SLDV), and epoxy potting. All estimates are used to predict the frequency-dependent cone velocities of three distinct loudspeakers. The actual velocities of their cones are measured under different conditions using the scanning laser, and each estimation is compared for accuracy. The estimation based on the SLDV blocked electrical impedance measurements proves to be the most precise in each case.

2010

Daniel Ross Tengelsen (Masters Thesis, August 2010, Advisor: Brian Anderson, Timothy Leishman )

Abstract

Two numerical techniques, the boundary-element method (BEM) and the finite-difference method (FDM), are used for simulating the radiation from horn-loaded compression drivers and from an infinitely-baffled, finite-length pipe. While computations of the horn-loaded compression driver are in steady state, transient analysis of the finite-length pipe is studied as a precursor to transient analysis within the horn-loaded compression driver. BEM numerical simulations show promise for the development of new designs. Numerical simulations serve as a good tool for time and cost-effective prototyping as poor designs are detected before they are built.

2009

Xi Chen (PhD Dissertation, December 2009, Advisor: Timothy Leishman )

Abstract

Equalization of loudspeakers and enclosed sound fields has been a topic of considerable interest for decades. Confusion has often arisen among audio professionals regarding the feasibility of simultaneously equalizing a loudspeaker and the enclosed field (i.e., the “room”) it excites. Because of frustrations encountered in such efforts, some have advocated abandoning simultaneous equalization altogether. This dissertation discusses the drawbacks of this approach as well as traditional in situ equalization methods. It demonstrates that many problems with traditional equalization stem from the use of measured acoustic pressure at a discrete point in a sound field as the system output. The dissertation presents analytical models and experiments involving the equalization of loudspeakers and both one-dimensional and three-dimensional sound fields. Equalization using total energy density at a point in either a one-dimensional or three-dimensional field produces better global equalization of the pressure field. In the onedimensional case, it allows simultaneous correction of spectral loudspeaker and global soundfield response anomalies in a nearly optimal sense.

Panu Tapani Puikkonen (Masters Thesis, April 2009, Advisor: Timothy Leishman )

Abstract

Sound pressure equalization of audio signals using digital signal processors has been a subject of ongoing study for many years. The traditional approach is to equalize sound at a point in a listening environment, but because of its specific dependence on the room frequency response between a source and receiver position, this equalization generally causes the spectral response to worsen significantly at other locations in the room. This work presents both a time-invariant and a time-varying implementation of an adaptive acoustic energy density equalization filter for a onedimensional sound field. Energy density equalization addresses the aforementioned challenge and others that relate to sound equalization. The theory and real-time implementation of time-invariant sound pressure and energy density equalizers designed using the least-squares method are presented, and their performances are compared. An implementation of a time-varying energy density equalizer is also presented. Time-invariant equalization results based on real-time measurements in a plane-wave tube are presented. A sound pressure equalizer results in a nearly flat spectral magnitude at the point of equalization. However, it causes the frequencies corresponding to spatial nulls at that point to be undesirably boosted elsewhere in the sound field, where those nulls do not exist at the same frequencies. An energy density equalization filter identifies and compensates for all resonances and other global spectral effects of the tube and loudspeaker. It does not attempt to equalize the spatially varying frequency nulls caused by local pressure nodes at the point of equalization. An implementation of a time-varying energy density equalizer is also presented. This method uses the filtered-x filter update to adjust the filter coeffi- cients in real-time to adapt to changes in the sound field. Convergence of the filter over time is demonstrated as the closed end of the tube is opened, then closed once again. Thus, the research results demonstrate that an acoustic energy density filter can be used to time-adaptively equalize global spectral anomalies of a loudspeaker and a one-dimensional sound field.

Brian Trevor Thornock (Masters Thesis, December 2009, Advisor: Timothy Leishman )

Abstract

"In room acoustics, the directional information of sound arrivals at a listening location can be used to diagnose the origins of problematic reflections so offending surfaces or other features can be properly treated. It can also be used for other purposes, including the study of psychoacoustic indicators. Many methods have been developed in the past to derive directional information, but despite their benefits, each has had significant drawbacks that have necessitated further research into their properties and development of an improved method. This thesis presents a review of past methods, their benefits and shortcomings. It discusses many theoretical and practical issues pertaining to the Polar ETC method and methods using the cross-correlation function. It also presents a new short-time correlation-based method (STCM) for gathering directional information of sound arrivals in rooms. Computer programs were developed for the implementation of the theory. Numerical simulations and experimental measurements are shown and the results are compared to those obtained by the Polar Energy Time Curve (ETC) method. The STCM is shown to be an improvement over past methods in terms of its ability to distinguish between simultaneous arrivals, its accuracy, its computational ef ficiency and its equipment requirements. Limitations of the method are also discussed.

2008

Ryan T Chester (Masters Thesis, March 2008, Advisor: Timothy Leishman )

Abstract

Several measurements may be used as error signals to determine how to appropriately control a sound field. These include pressure, particle velocity, energy density and intensity. In this thesis, numerical models are used to show which signals perform best in two situations. The first is free-field active noise control (ANC) using error sensors located in the near field of the sound sources. The second is equalization in a free field and a semi-free field. Minimized energy density total power output (MEDToPO) plots are developed; these indicate the maximum achievable attenuation for a chosen error sensor as a function of location. A global listening area equalization coefficient (GLAEC) is found to evaluate the performance of the equalization methods. It is calculated by finding the average of the spectral standard deviation of several frequency response measurements in a specified listening area. For free-field ANC employing error sensors located in the near field, pressure-based measurements perform the best. For free-field equalization over an extended listening region, total energy density performs best. Equalization of an extended listening region is more successful over a limited low-frequency bandwidth.

Nathan Curtis (Capstone, August 2008, Advisor: Timothy Leishman )

Abstract

A new method for obtaining and calculating directional squared impulse re- sponses (IRs), and hence locating re ective surfaces in rooms, is assessed for patentability purposes. It is compared to other methods of obtaining direc- tional impulse response measurements, including the polar energy-time curve method. It is determined that the new method passes three requirements for patentability but fails the test of unobviousness. A draft patent for this method is included in case patentability is determined more favorably in the future.

Michael Mendoza (Capstone, August 2008, Advisor: Timothy Leishman )

Abstract

A brief history of anechoic chambers is presented. The impedance tube method outlined in the ASTM standard E1050 was used to validate and optimize a double-tipped foam wedge construction for a new anechoic chamber. Various foam densities were tested and a relatively high density was found to work. After installing these wedges into the chamber, it was further qualified using the procedure outlined in ISO standard 3745. These results were compared to the preliminary tests. Further chamber construction details are outlined, including the design of foam wedge door, a wire mesh floor, and the foam wedge support structure used to attach the wedges to the chamber walls.

Daniel Tengelsen (Capstone, August 2008, Advisor: Timothy Leishman )

Abstract

A regular polyhedron loudspeaker (RPL) is a sound source used by architectural acousticians to approximate an omnidirectional source. It is composed of a polyhedron enclosure (i.e., tetrahedron, hexahedron, dodecahedron, etc.) that is fitted with a loudspeaker driver in each face. Each face provides a distinct axis in which the RPL can radiate sound. At low frequencies, an RPL approximates an omnidirectional source. As frequency increases, the radiated sound begins to lobe in a multidirectional pattern. Measurements have shown that the omnidirectionality of most RPLs becomes poor above 1 kHz, which is well below the typical maximum frequency of interest. This presentation reports preliminary theoretical and experimental work aimed to better understand how the geometry of a loudspeaker driver affects an RPL's omnidirectional behavior. It also includes corollary evidence between the interstitial area of the RPL and its total omnidirectionality.

2007

Matthew Morrise (Senior Thesis, August 2007, Advisor: Timothy Leishman )

Abstract

Equalization is often used to optimize the frequency response of an audio system in a listening environment. However, no standardized metric exists for quantifying the performance of equalization filters. This paper introduces the concept of global equalization, or equalization for multiple listening positions, and reviews several methods that might be used to quantify the related performance of equalization schemes. It then introduces a new metric, the equalization quality coefficient, and tests it against other metrics. Discrepancies between rankings provided by the equalization quality coefficient and expected results are analyzed. Another metric, the spectral standard deviation of the spatially averaged time-mean-square pressure, is contrasted with the equalization quality coefficient. The author concludes that the equalization quality coefficient provides the best evaluation of equalization performance.

Brian Thornock (Capstone, April 2007, Advisor: Timothy Leishman )

Abstract

In room acoustics, it has been common practice for many years to characterize a room using impulse response measurements of the room taken from only a few locations. One of the major problems with this method is that a few measurements in even a medium sized space are generally insufficient to describe what is happening throughout the hall with acceptable accuracy. With the advent of multi-channel software based analyzers, a need has been created for a circuit that will allow the use of current industry-standard precision measurement microphones with new multi-channel input interfaces. Several circuits were designed to convert phantom power to ICP power and single ended signals into differential signals. The three best designs were chosen for prototyping. After analysis of the prototypes, the two best circuits were chosen. Prototypes underwent preliminary testing to ensure functionality. The prototypes were then compared to an ICP microphone connected to a signal conditioner. Frequency response, noise floor and relative total harmonic distortion (THD) were measured and presented graphically. Both of the designs were found to match the performance of the ICP microphone and signal conditioner very well. A final recommendation was made based on simplicity of circuitry, component count and types of components used.

2006

Gordon Robert Dix (Masters Thesis, May 2006, Advisor: Timothy Leishman )

Abstract

Highly directive loudspeakers have long been important tools for sound system designers, experimental acousticians, and many other professionals in the audio industry. They allow sound engineers to more easily manipulate the radiation pattern of their loudspeakers to accommodate the purpose of the venue. Many commercially available products, while exhibiting good directivity at mid and high frequencies, generally lack control in the low frequency range. A new method for controlling the radiation pattern of a loudspeaker at low frequencies has been developed and modeled extensively. Prototypes have been built and tested in an anechoic chamber. Results from computer modeling and experimental measurements will be presented and compared in this thesis.

Shinji Inagi (Capstone, August 2006, Advisor: Timothy Leishman )

Abstract

The description of the sound quality of organ pipes has received little attention either in organ-building or scientific literature, despite its importance in organ pipe voicing. Voicing of organ pipes is considered as art and there is no set standard method. Thus a voicer has to rely on his or her own ears, experience, skills, and talents in the voicing process. Is this research, I have analyzed the effect of the upper lip of an organ pipe on its sound quality. I focused on how the shapes and geometrical proportions of the upper lip of an organ pipe affect the sound. The results are shown with unique peaks of harmonics on frequency-domain graphs.

David B Nutter (Masters Thesis, August 2006, Advisor: Timothy Leishman )

Abstract

Measurements in a reverberation chamber use spatially averaged squared pressure to calculate sound absorption, sound power, and other sound measurements. While a reverberation chamber provides an approximation of a diffuse sound field, variations in the measurements introduce uncertainty in measurement results. Room qualification procedures require a sufficient number of source-receiver locations to obtain suitable measurements. The total acoustic energy density provides greater spatial uniformity than squared pressure, which requires fewer source-receiver positions to produce similar or better accuracy in measurement results. This paper explores the possibility of using energy density in place of squared pressure, using methods outlined in current ISO standards, by describing several experimental and analytical results.

Brandon Skinner (Capstone, December 2006, Advisor: Timothy Leishman )

Abstract

A new system for measuring the vibrations of metal strings in musical instruments is devised and experimented on. A set of matched magnetic transducers are produced as well as a test apparatus for holding a vibrating string. The pickups are mounted close to the string in a 2-dimensional array. Measurements of said apparatus are taken using the magnetic pickup array and analyzed using FFT analysis in MATLAB. These results are discussed quantitatively and qualitatively, and a discussion of the pros and cons of such an approach to transduction is given.

2005

Steven Davis (Capstone, July 2005, Advisor: Timothy Leishman )

Abstract

Experimental procedures are developed to measure the three-dimensional directivity patterns of large acoustical sources using a single arc array of 19 microphones. A computer-controlled turntable is utilized to enable measurements at various azimuthal angles. Methods are instituted to ensure normalized patterns for time-varying signals. Measurements and three dimensional balloon plots are included to verify these methods.

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.

Connor Duke (Capstone, April 2005, Advisor: Timothy Leishman )

Abstract

A high-fidelity loudspeaker system was made within a specific budget. This system is compatible with current 5.1 surround encoding systems and future 6.1 and 7.2 surround systems. It meets basic THX standards by having seven full range (50 Hz – 20 KHz), three-way surround speakers and two low-frequency effects speakers. These criteria are met while keeping the system aesthetically pleasing.

Bryan Keeler (Capstone, November 2005, Advisor: Timothy Leishman )

Abstract

n/a

Sarah Rollins (Masters Thesis, November 2005, Advisor: Timothy Leishman )

Abstract

In order to preserve the acoustics of the Salt Lake Tabernacle after the seismic renovation of 2005-2006, it was necessary to characterize these acoustics immediately preceding the renovation. This thesis discusses the characterization process that began with the measurement of hundreds of impulse responses for five different source positions and several receiver locations throughout the hall seating areas. The acoustics were further characterized by deriving various parameters from these responses that correlate with subjective preferences for music and speech. Impulse responses were also generated by a CATT-Acoustic™ computer model of the Tabernacle for the same purpose. The parameter values were then mapped over diagrams of the seating areas of the hall to show the spatial variation of the acoustics. To further investigate the variation, statistics were calculated for each parameter and an algorithm was developed to determine the minimum number of receiver locations necessary to adequately characterize the hall. Computer models were also used to investigate focusing effects of the curved ceiling and historical comments made about the improvements to the acoustics with addition of the balcony in 1870.

Benjamin Shafer (Capstone, August 2005, Advisor: Timothy Leishman )

Abstract

This report provides a summary of the procedure used to qualify a reverberation chamber for the purpose of absorption and sound power measurements. The physical qualities of a diffuse sound field are discussed as well as mathematical descriptions of the characteristics of a reverberation chamber and the sound absorption of materials therein. The reverberation chamber structure and the fabrication and placement of sound diffusers are described and compared to similar experiments for qualification. The experimentalists cited place different numbers of sound diffusers in the reverberation chamber and use various methods of achieving a diffuse sound field in the chamber. Their methods are discussed and compared. The reverberation chamber is qualified according to international standards using a 12 microphone measurement for each frequency in the one-third octave bands ranging from 100 to 5000 Hz.

Sentagi Sesotya Utami (Masters Thesis, August 2005, Advisor: Timothy Leishman )

Abstract

Concave surfaces are often considered to be detrimental or precarious in room acoustics, especially because of the impact they have on the distribution of sound energy. However, it is often difficult to avoid such surfaces in buildings with specific architectural functions. A primary example of this involves mosques, which are sacred places of worship for Muslims. In keeping with the Islamic architectural style, most mosques incorporate a symbolic centralized domed ceiling as part of their roof structures. These domes are open on the bottom and coupled to the acoustic spaces below. In many cases, the lower spaces may be idealized as rectangular enclosures. Owing to the distinctness and ubiquity of this basic architectural form, a thorough, fundamental analysis of such environments would be useful to the architectural acoustics community. In this study, predictions from EASETM computer models were compared to the results derived from physical scale model measurements. The scale model measurement techniques involved evaluation of impulse responses in a 1:12 scale model of Darussholah mosque, in East Java, Indonesia. A miniature human voice source was created to carry out the impulse response measurements. It was carefully evaluated to ensure that it produced adequate frequency response and directivity comparable to an actual human voice. Acoustical parameters were derived from the impulse responses. Statistical analysis using ANOVA and t-tests were used to compare results from the measurements with variations of domed ceiling configurations and other aspects of the measurement setting. Conclusions were based on these comparisons and on auralization listening tests in order to ascertain the elements that produced the most significant impact on the mosque acoustics. The analysis helps establish criteria for good acoustics in mosques and other buildings with domed ceilings.

2004

Brian E Anderson (Masters Thesis, January 2004, Advisor: Timothy Leishman )

Abstract

Small-signal moving-coil loudspeaker driver parameters are traditionally derived through electrical impedance measurement techniques. These parameters are commonly called Thiele/Small parameters, after Neville Thiele and Richard Small who are credited with developing industry-standard loudspeaker modeling techniques. However, because loudspeaker drivers are electro-mechano-acoustical transducers, it should be possible to measure their parameters in physical domains other than the electrical domain. A method of measuring loudspeaker parameters from the acoustical domain will be developed. The technique uses a plane wave tube to measure acoustical properties of a baffled driver under test. Quantities such as the transmission loss through the driver are measured for a driver placed in the tube using the two-microphone transfer-function technique. Models have been developed to curve fit the resulting data, from which small-signal loudspeaker parameters are subsequently derived. This thesis discusses the acoustical measurement theory, apparatus, and system modeling methods (via equivalent circuits). It also compares measured parameters to those derived using electrical techniques. Parameters derived from both approaches are compared with reference values to establish bias errors. Sequential measurements are also compared to reveal random errors in the derivation processes.

Ryan Chester (Senior Thesis, December 2004, Advisor: Timothy Leishman )

Abstract

n/a

Kara Kemsley (Senior Thesis, April 2004, Advisor: Timothy Leishman )

Abstract

n/a

Heather Smith (Masters Thesis, November 2004, Advisor: Timothy Leishman )

Abstract

This thesis discusses the process of modeling a 21,000 seat fan‐shaped auditorium using methods of geometric acoustics. Two commercial geometric acoustics software packages were used in the research: CATT‐AcousticTM 8.0 and EASETM 4.1. The process first included creating preliminary models of the hall using published absorption coefficients for its surfaces and approximate scattering coefficients based on current best‐known techniques. A detailed analysis determined the minimum numbers of rays needed in both packages to produce reliable results with these coefficient values. It was found that 100,000 rays were needed for CATTTM and 500,000 rays were needed for EASETM. Analysis was also done to determine whether the model was sensitive to the scattering coefficients of the seating areas. It was found that most acoustic parameters were not significantly affected by scattering coefficient variation. The models were subsequently refined by including measured absorption coefficients of dominant surfaces in the hall: the seats, audience and suspended absorptive panels. Comparisons were made between measurements made in the hall and results from the computer models with impulse responses, acoustic parameters, and auralizations. The results have shown that the models have been successful at representing characteristics of the hall at some positions but less successful at representing them at other positions. Comparisons have shown that positions on the rostrum were especially difficult positions to model in this hall. Significant differences were not found between the preliminary models and the refined models. There was not significant evidence showing that either the EASETM or the CATTTM model was more successful in accurately representing the acoustical conditions of the hall. The results from this research suggest that more work must be done to improve the modeling capabilities of these packages for this application.