bluuebunny/crossref_metadata_2025 huggingface dataset:
Hybrid methods for incorporating density variations into the split-step PE
Authors: Yevick David , Thomson David J.
Published: October 2005

Density variations are easily analyzed in the context of finite difference parabolic equation (PE) solvers by discretization of an appropriate differential operator. In split-step Fourier solution algorithms, however, variations in the density ρ are instead modeled by adding terms to the refractive index. Since these extra terms depend on derivatives of ρ, geoacoustic density profiles must be smoothed appropriately to remove any step discontinuities. In this paper, a new hybrid method is proposed for treating density inhomogeneities in the split-step PE. This approach involves splitting the differential operator into density-independent and density-dependent components. While the former component is propagated using the split-step Fourier technique, the influence of density changes is computed through a finite difference procedure. Such an algorithm is especially attractive as it may be transparently incorporated into the recently proposed hybrid split-step/finite-difference and split-step/Lanczos solvers [J. Acoust. Soc. Am. 96, 396–405 (1994)]. Both the hybrid and the standard finite-difference procedures are applied to a shallow-water test case involving jump discontinuities in both the sound speed and the density and are found to be in excellent agreement with reference solutions.
Report on the Cerritos Centre for the Performing Arts
Authors: Godden Jerry
Published: October 2005

Small and medium sized communities wanting a high quality performing arts space are faced with reconciling the conflicting demands that different arts forms make on the performance space. The recently completed Cerritos Centre for the Performing Arts in California is unusual in its design in that it utilizes modern technology to provide a space for the performing arts that can become a 1896-seat concert hall, a 1450-seat lyric theatre, a 900-seat drama theatre, a 1934-arena theatre, or a cabaret space seating 1472. The auditorium is transformed including its seating, its sightlines, and its acoustics by the use of movable seating towers. The design demonstrates how a small town can afford to build and operate a facility in scale with its capabilities and aspirations and may well prove to be a key point in the concept of multi purpose spaces of the future.
Acoustic project of ‘‘Mexico Radio ABC’’ broadcasting rooms
Authors: Valdés Ana Maria , Vázquez-Raña Mario
Published: October 2005

Design for room acoustic insulation, reverberation time, and homogeneous diffusion of acoustic field, results in high building budgets for broadcasting and TV stations, theaters, etc., projected in Mexico. For this reason, they are often constructed without an adequate acoustic design. This paper presents the acoustic design applied to the construction of three broadcasting studios for ‘‘Mexico Radio ABC,’’ considering both a low cost project and long life materials fitted for the Mexico City’s environment, and taking into account a new national draft standard. The project was planned for noisy surroundings (sound pressure levels averaged of 78 dB, A-weighted). External noise comes mainly from traffic, nearby schools and helicopters; indoor noise comes from air conditioned and background office noise. The original volume shape was a parallelepiped, which had to be transformed to avoid undesirable reflections. The reverberation time chosen for control rooms was 0.35 s at 1000 Hz, and 0.22, 0.25, and 0.35 s at 1000 Hz for broadcasting studios; these values were achieved in practice. Final costs were low, in comparison with a construction without previous acoustic design.
Sound isolation of an audiometric test room
Authors: Bommer Arno S. , Moritz Charles T.
Published: October 2005

Modifications were required to reduce sound levels in a former office so that the room would meet the ambient noise requirements for audiometric testing. Sound sources in the area included people in neighboring offices, transformers in a neighboring mechanical equipment room, ventilation ducts in the plenum above the room, and air-conditioning vents. The transformers created harmonically related vibration tones in the concrete floor slab and walls that radiated sound into the proposed audiometric test room. Acoustical recommendations consisted primarily of building a vibration-isolated room within the existing room (including a floating floor) and acoustically treating the ventilation system. Initial measurements revealed unexpectedly high sound levels that were the result of unintended (erroneous) structural connections between the inner and outer room. When these connections were cut, the room met the noise requirements.
BACH, a new binaural room acoustical measure
Authors: Carvalho António Pedro O.
Published: October 2005

This study reports on acoustical field measurements made in a major survey of 41 Catholic churches in Portugal built in the last fourteen centuries. Binaural measurements were taken in each church using a dual-channel real-time frequency analyzer to calculate the coherence values between the signals at both ears in 1/3-octave frequency bands. From the coherence values a new binaural acoustical measure was developed and called BACH (BinauralAcousticalCoHerence). Monaural acoustical measurements were also taken at several source/receiver locations using the impulse response method. BACH was shown to be an orthogonal parameter in statistical tests with nine other monaural acoustical measures (RT, EDT, C80, D, TS, L, BR—RT, BR—L, and RASTI) and fifteen architectural parameters. Information was collected regarding the quality of music in each church. ANOVA tests were performed to examine the significance of the differences among the groups of equal subjective quality ratings. The author wanted to test the hypothesis that this new binaural acoustical measure can be useful in estimating the general subjective quality of churches regarding music. A linear correlation coefficient near 0.7 was found between the BACH and the subjective quality ratings that supported the stated hypothesis. This is exploratory research and developments are being tested to improve the model. [Work supported by JNICT/Ministry of Planning and Univ. of Porto, Portugal.]
Experimental results from hybrid control with a sensoriactuator
Authors: Clark Robert L. , Vipperman Jeffrey S.
Published: October 2005

Both transient and persistent disturbance rejection were demonstrated experimentally on a cantilevered beam configured with a piezoelectric sensoriactuator. The transient response of the system was suppressed through direct-rate feedback control, and adaptive feedforward control was utilized to minimize the response to a harmonic input disturbance. A time-averaged gradient descent algorithm was implemented to adapt a finite impulse response filter in the feedforward control approach. Experimental results demonstrate that rate-feedback control can be utilized to enhance the transient adaptation of the feedforward control algorithm. Furthermore, the sensoriactuator provides a convenient method of performing both sensing and actuation simultaneously in feedback and feedforward control of adaptive structures.
Determination of effective secondary source locations for active noise control
Authors: Yang Jihe , Holger David K.
Published: October 2005

A numerical method for determining effective secondary source locations for active control of interior sound fields has been investigated. The method uses intermediate results from an indirect boundary element simulation of a sound field to determine effective boundary locations for secondary sources. In the indirect boundary element method (IBEM), an interior sound field is simulated by replacing the physical boundaries with a fictitious source distribution that is determined from the geometry, the properties of the physical boundaries, and the primary source location(s). Locations of high fictitious source strength, as determined by the IBEM, are found to be particularly effective locations for secondary sources that are components in three dimensional active noise control systems. Numerical results for simple geometries are in agreement with previous experimental results [Elliott etal., J. Sound Vib. 117, 35–58 (1987)], and numerical predictions of active noise control using the proposed method for locating secondary sources resulted in sound pressure level reductions of more than 20 dB in reverberant and semi-reverberant spaces. The results obtained suggest that the method has significant potential for efficiently locating effective secondary sources for a variety of active noise control applications.
Microphone arrays for reducing reverberation and noise in speech communication
Authors: Ryan James G.
Published: October 2005

Microphone pickup of sound in typical rooms is impaired by the combined effects of reverberation and noise. This degrades speech intelligibility and quality particularly in applications where the microphone is located far away from the talker. Recent advances in microphone array technology suggest a potential solution to such problems. This paper gives an overview of current microphone array techniques and discusses the potential benefits for speech communication. Various criteria for measuring the performance of a microphone array are described. A flexible, experimental microphone array intended for research in speech communications is under construction and will be described.
Binaural arrays for hearing enhancement
Authors: Scanlon Michael V. , Tenney Stephen M.
Published: October 2005

Two hearing augmentation devices developed at the Army Research Laboratory can enhance normal listening abilities and restore hearing degraded by encapsulating headgear. Surrounding sounds are localized with a head-mounted binaural pinna attachment that recreates the head-related transfer function associated with the normal listening. The user’s brain interprets the recreated stereo signals that enter the ear canals through intra-aural speakers, giving excellent restoration of omnidirectional hearing. A hand-held, ultra-directional array extends the user’s listening range. The use of delay and sum beamforming in the array assures maximum directivity in the pointing direction. The binaural long-range hearing device has two linear endfire arrays of eight cardioid microphones each. The slightly offset directivity patterns of the two arrays create stereo outputs, so that the user can interpret differences in amplitude, phase, time-of-arrival, and frequency content of sounds in the forward area. These devices provide aural protection and an intra-aural input for communications, without removing the user from his acoustic environment. Both devices can be monitored remotely, and are ideally suited for detecting speech, personnel, equipment, or vehicles during military or law enforcement missions. Performance measurements of various array configurations will be shown.
Field impact insulation class (FIIC)—A case study
Authors: LoVerde John J. , Mange Gary
Published: October 2005

Impact noise in buildings constitutes a potentially serious problem because of the short duration, high intensity sounds involved [U.S. Department of Housing and Urban Development, Airborne, Impact, and Structure Borne Noise, Chapter 7 (1967)]. Since 1974, the State of California has tried to manage this problem by instituting the California Noise Insulation Standards, which require any multi-family dwelling to provide an impact insulation class (IIC) rating of 50 based on laboratory tests, or a field impact insulation class (FIIC) rating of 45 based on field tests [Office of Noise Control, California Noise Insulation Standards, 1–7 (1988)]. Concern about the acceptability of floor ceiling assemblies is increasing due to increased awareness of the problem and larger numbers of people moving into apartments, condominiums and townhomes throughout California. Western Electro-Acoustic Laboratory (WEAL) had the opportunity to witness the installation of floor ceiling assemblies in an apartment complex in Bakersfield, California. Six different assemblies were tested to determine how the FIIC value changed when minor modifications were made to the standard floor ceiling assembly. WEAL will show the results of the field-tested assemblies and compare the data with typical laboratory results for similar constructions.
Effect of concave sound reflecting surfaces on speech intelligibility and the articulation index
Authors: Khaiyat Sami A. , Boyer Lester L.
Published: October 2005

Two different methods, the calculation of the articulation index (AI) and the rapid speech transmission index (RASTI) measurement using the speech transmission meter, were utilized to obtain the speech intelligibility in spaces with concave sound reflecting surfaces. Many factors were considered such as room size, size of curvature, position of the sound source, and background noise level. The base cases, spaces without curvatures, showed highly correlated results indicating no significant differences between calculated AI and measured RASTI. With curvatures and the sound source at 4 ft. away from the center position of the front wall it was found that the effect of room size is significantly different between the two methods under the 2- and 16-ft. radius curvatures only; also, the background noise level showed almost the same effect on both calculated AI and RASTI. Under low background noise levels, spaces with curvatures showed that the effect of change in room size for both methods is identical. The center location for the sound source showed better speech intelligibility under all different testing conditions. Finally, a modification factor is developed and applied to the calculated AI so that reliable estimates of speech intelligibility in spaces with curvatures may be obtained.
An aspect of sonoluminescence from hydrodynamic theory
Authors: Kwak H. , Yang H. , Hong J.
Published: October 2005

Sonoluminescence (SL), the phenomenon of light emission associated with the collapse of bubbles oscillating under an ultrasonic pressure field has been studied by solving the continuity, momentum (Euler), and energy equations for the gas inside the bubble analytically. Heat transfer in the liquid layer adjacent to the bubble wall has also been considered in this analysis. It has been found that the gas behavior is neither adiabatic nor isothermal for a bubble under ultrasound conditions. In this analysis, the launch condition and the Hugoniot curve for the shock propagation has been identified, and the shock duration of 2.7 to 17 ps, which is comparable to experimental results, has been obtained with the help of a similarity solution (Guderley) for converging spherical shock. For SI, the gas temperature after the shock focusing has been found to be 7000–44 000 K, depending on the equilibrium bubble radius and the driving amplitude of ultrasound. It has also been found that the heat flux at bubble collapse is as large as 47 GW/m2, which could be more than enough to cause an explosion of an explosive crystal.
On the use of probe microphone and level difference measurements to characterize air-filled porous media
Authors: Frederickson Carl K. , Sabatier James M.
Published: October 2005

Air-filled porous media have been characterized acoustically using level difference measurements and an impedance model that depends on a pore-shape factor, porosity, tortuosity, and bulk flow resistance. Least-squares fitting of level difference spectra only allows two of the above parameters to be independently determined. Probe microphone measurements have been used to determine the propagation constant in both washed sand and glass beads. Values of tortuosity calculated from probe microphone measurements were used in the analysis of level difference data to determine porosity and bulk flow resistance. An average pore-shape factor is used in the analysis. For the unconsolidated porous media used, the effect of the pore-shape factor variation was within the error range of the measurement. The availability of both probe microphone and level difference data has also allowed for the comparison of porosity and flow resistance values calculated from each set of data. There are some discrepancies between the frequency dependence of the model and the probe microphone data. [Work supported by USDA.]
Acoustic probe microphone measurements of Biot type I and II waves in air-filled sands
Authors: Hickey Craig , Prather Wayne , Sabatier James M.
Published: October 2005

Probe microphone measurements of air-borne sound penetrating into air-filled sands and soils indicate two absorption coefficients for the frequency range 40–4000 Hz. The probe microphone signal attenuates rapidly with depth in a region near the surface. Below that region the microphone signal attenuates significantly slower with depth. Rigid-capillary-tube porous models, which allow for pore-fluid motion only, correctly describe the rapid attenuation of probe pressure. Using these models, pore properties (tortuosity and air permeability) are typically deduced from the measured complex absorption coefficient. The Biot poro-elastic-capillary tube model describes both attenuation regimes. The two absorption coefficients are associated with the Biot type I and II waves. The large attenuation in the region near the surface is associated with the Biot type II wave. The much smaller attenuation coefficient is a consequence of the elasticity of the matrix and is associated with the Biot type I wave. Biot’s model is used to calculate the microphone pressure from both Biot type waves as a function of depth. [Work supported by USDA.]
The influence of pore-size distributions on complex wave number in air-filled porous materials
Authors: Craig David W. , Frederickson Carl K. , Sabatier James M.
Published: October 2005

Probe microphones have been developed to determine tortuosity and effective flow resistivity for air-filled porous materials, such as agricultural soils. They are used to measure complex wave number as a function of frequency. These data are then inverted using a single-pore-size capillary tube model for propagation in the material. However, the frequency dependence of the measured wave number in sand differs from the predictions of such single-scale models. It is shown that distributions of pore sizes can produce a similar dependence. By summing over pore sizes, wave number as a function of frequency is calculated for lognormal, fractal (power-law), and empirical distributions derived from porosimetry data. Results are also compared with Wilson’s relaxation-matched fractal model [D. K. Wilson, J. Acoust. Soc. Am. 94, 1136–1135 (1993)]. [Work supported by USDA.]
Sound propagation in capillary-tube-type porous media: Effects due to the presence of absorbed water in the capillary walls
Authors: Bernard Miguel , Sabatier James M.
Published: October 2005

The effects on sound propagation is investigated in air-filled capillary-tube-type ceramic porous media that result from the presence of small quantities of absorbed water in the tube walls. Specific acoustic impedance measurements are performed for a rigid-backed sample for the cases of both porous and nonporous tube walls. For the nonporous tube walls measurements are performed on dry and water wetted walls. Preliminary results suggest a contribution to the increased attenuation of sound in the air-filled porous samples due to the presence of a thin water film. [Work supported in part by ONR.]
Modeling sound propagation in a high porosity fibrous material
Authors: Morfey Christopher L. , Semeniuk Bradley P. , Petyt Maurice
Published: October 2005

Sound propagation in an air-filled, high porosity fibrous material involves the elastic response of the fiber skeleton as well as thermal and viscous effects at the fiber–fluid boundaries. A theoretical model of wave propagation in such a medium has been constructed based on the idea of an equivalent fluid, which occupies the entire space and whose properties approximate the locally averaged properties of the actual fluid. Once the elastic properties of the skeleton are known—e.g., from invacuo experiments—the model yields phase speeds, attenuation rates, and complex characteristic impedances for both fast and slow waves in the composite medium. Typical results are presented for fiberglass blankets of the type used as thermal insulation in aircraft fuselages. [BRAIN project supported by the European Commission.]
Sound transmission in a pipe with developing laminar flow: Upstream/downstream phase speed differences
Authors: Morfey Christopher L. , Smith Malcolm G.
Published: October 2005

In a uniform rigid-walled duct, plane waves propagate with axial phase speed (1±M) times the sound speed, according to the inviscid plug-flow model. The difference between the downstream and upstream phase speeds provides a measure of the flow rate; the same concept can be extended to realistic profiles of M (Mach number) across the duct section, although the lowest-order mode is not a plane wave any more. Calculations were carried out to provide a means of converting phase speed differences to flow rates, assuming that propagation is confined to a single low-order mode (i.e., near-axial propagation). The results were then compared with phase speed measurements made in a cylindrical steel tube, through which air was pumped at a controlled steady flow rate. Close but not perfect agreement was found, which raises the question of whether our neglect of thermoviscous phenomena is justified, particularly near the duct walls. [Work supported by British Gas.]
Wavelet analysis of transient-evoked otoacoustic emissions
Authors: Travis Jeffrey D. , Thornhill R. Joe
Published: October 2005

A novel approach to finding information in transient-evoked otoacoustic emissions (TEOAEs) was used: the discrete wavelet transform. It is argued that wavelet transforms are more appropriate than fast Fourier transforms (FFTs) for signal decomposition because of the transient, nonstationary nature of TEOAEs. Several wavelets were used to obtain time–frequency maps of TEOAEs collected in past experiments. The latency of several frequency bands can be clearly determined for any given subject with this map. Another set of data collected from an experiment in which subjects were administered quinine sulfate was also studied. Quinine sulfate suppresses TEOAEs and induces some hearing loss [D. McFadden and E. G. Pasanen, J. Acoust. Soc. Am. 95, 3460–3474 (1994)]. Wavelet analysis on this data revealed any temporal shifts of frequency bands during the effect of the drug, and correlated the TEOAE waveform with the amount of temporary hearing loss at each frequency band.
Effect of anticholinergic drugs on human spontaneous and click-evoked otoacoustic emissions, frequency discrimination, and lateralization using interaural intensity differences
Authors: Callaway Narriman Lee , McFadden Dennis
Published: October 2005

The primary neurotransmitter for the cochlear efferent system is widely believed to be acetylcholine. If anticholinergic drugs reduce neurotransmitter activity in the efferent pathway, they should produce effects in accord with those seen in animal studies where the olivocochlear bundle (OCB) was severed. Experiments were conducted to measure effects of three common anticholinergic drugs, diphenhydramine (BenadrylR), hyoscyamine (LevsinR), and scopolamine (Transderm-ScopR) on two psychophysical tasks and two physiological measures. There was no evidence of impaired performance in a complex frequency-discrimination task or in a lateralization task using interaural intensity differences. Also, no compelling evidence was found to indicate that either spontaneous or click-evoked otoacoustic emissions were enhanced by these drugs. After the beginning of these experiments, a report appeared [S. G. Kujawa etal., Hear. Res. 74, 122–134 (1994)] indicating that, in guinea pigs, suppression of distortion-product otoacoustic emissions was reversed by anticholinergics having antinicotinic action, but less so by those having antimuscarinic action. The three anticholinergic drugs used here are primarily antimuscarinic in their action, which may explain the failure to observe the predicted effects. [Work supported by NIDCD.]
Individual differences in sensitivity to interaural disparities of time and of level
Authors: Langford Ted L.
Published: October 2005

The sensitivities of 22 women and 23 men to interaural temporal differences (ITDs) in low-frequency noise bursts and interaural level differences (ILDs) in high-frequency noise bursts were measured in discrimination and absolute judgment tasks. The discrimination abilities of both groups of observers were symmetrical in that there were no differences in performance between interaural disparities favoring either the right or the left ear. The women were somewhat less sensitive and more variable in their performances than were the men in both the temporal and level difference discrimination tasks. However, the mean differences between the sexes, 28 μs and 0.9 dB, were not large compared with the variability across individuals. Individuals who performed well-discriminating ILDs also performed well with ITDs. Many observers exhibited ILD subjective midlines which were offset to the left of objective center in the absolute judgement task. This effect was unrelated to ILD discrimination ability, to monaural thresholds, or to absolute judgements based on ITDs.
Contralateral stimulation and the masking overshoot effect
Authors: Turner Christopher W. , Waite Rebecca , Cummings Karolyn , Rosen Melanie
Published: October 2005

Overshoot is the increase in masked threshold for a short signal presented at the onset of a masker compared to the threshold for a signal presented in the temporal center of the masker. One hypothesis to explain this effect is the slow onset time for masker-stimulated ipsilateral efferent activity to influence the response to the masked signal. Recent physiological experiments have shown that noise presented to the contralateral ear, which provides an increment in efferent activity, can increase the neural detectability of short tones masked by noise [Kawase etal., J. Neurophys. 70, 2533–2549 (1993)]. The present behavioral experiment took advantage of these findings and measured the amount of overshoot under conditions where a brief contralateral noise was presented prior to the masker onset, in an attempt to ‘‘prime’’ the efferent system prior to the masker onset. All measurements were conducted using insert earphones, which provide approximately 80 dB of interaural attenuation. The preceding contralateral noise reduced or eliminated the overshoot across a wide range of masker levels. Possible mechanisms for the overshoot effect will be discussed. [Work supported by NIDCD.]
A computational model of temporal integration
Authors: McAngus Todd Neil P.
Published: October 2005

In this paper a computational model of temporal integration is demonstrated. The model has the following architecture. The first stage computes a representation of the auditory nerve response based on the Sheffield ear. The nerve response is summed and low-pass filtered with a temporal window of about 10 ms [Plack and Moore, J. Acoust. Soc. Am. 87, 2178–2187 (1988)]. In the second stage the summed and smoothed nerve response is coupled to a transmission line model of auditory sensory memory. Close to the periphery the impulse response of the transmission line is sharp. As a pulse transmits into the system though, it becomes progressively more attenuated and spread out in time. In the third stage the line is tapped at different points (approximately 10-ms spacing) and peak responses detected, which then become input samples for higher order leaky integrator with a time constant of about 200 ms. The model thus resembles the ‘‘multiple looks’’ model as proposed by Viemeister [Viemeister and Wakefield, J. Acoust. Soc. Am. 90, 858–865 (1991)] and is able to account for the ‘‘resolution-integration’’ paradox. Quantitative predictions of the model are shown for the temporal integration example from the Houtsma, Rossing, and Wagenaars compact disk.
Signal processing to improve speech quality for hearing-impaired people
Authors: Ribic Zlatan , Yang Jun
Published: October 2005

A signal processing scheme is proposed to improve speech quality for the hearing-impaired people (HI). In this scheme, Hilbert transform and vector sum generator are used to obtain the Hilbert envelope AM, and it follows a voltage-controlled amplifier (VCA) with the gain inversely proportional to AM, the output signal of VCA is sent to a modulator. On the other hand, AM is dynamically processed by a lowpass filter and nonlinear amplitude processor, then sent to the said modulator to restore the dynamic range of the speech. In comparison with the original speech, the resulting one is of a little spectral enhancement, the intermodulation-distortion (ID), and some compression effect because of the nonlinear amplitude processing. Towards the end of our research, the informal listening experiment was performed, 25 people with sensorineural hearing impairment who wear their own hearing aids routinely participated in the experiment. The experiment results have shown that 80% listeners preferred the processed speech to the original one. This result is quite consistent with one of our expectations: HIs lose the nonlinear active characteristics to some extent, therefore, introducing appropriately ID can help HIs to perceive speech.
Acoustic characteristics of voiceless plosives: For the improvement of consonant recognition ability in cochlear implant
Authors: Omori Koichi , Blaugrund Stanley M. , Kojima Hisayoshi , Shoji Kazuhiko
Published: October 2005

Recognition of consonants, especially voiceless plosives, is inadequate in patients who undergo multichannel cochlear implants. This study was undertaken to ascertain acoustic characteristics of voiceless plosives /p/, /t/, and /k/. Consonant-vowel syllables /pa/, /ta/, and /ka/ of a normal human voice were digitized, and processed by computer in three different ways. First, a portion of consonant signals was deleted. Second, a preceding consonant in one syllable and a following vowel in the other syllable were combined. Third, vowel /a/ was synthesized by repetition of the first cycle of following vowels, and was examined by spectral analysis. All processed sounds were perceived by five experienced listeners. Each voiceless plosive was recognized correctly by its processed syllables which had at least a 1-ms signal of consonant from the onset. Following vowels served as a cue for recognizing voiceless plosives. In following vowels, frequency information, especially in the high-frequency range, was an important factor. From these results, recognition of a short signal at the onset of preceding consonants and emphasis of high-frequency power in following vowels are essential for the improvement of speech perception of voiceless plosives in cochlear implantation.
Active control of flexural and extensional power flow in beams using time domain wave vector sensors
Authors: Gibbs Gary P. , Fuller Chris R.
Published: October 2005

Active control of beam vibration using power flow considerations has been a topic of interest in recent years. Traveling waves in structures can be considered ‘‘carriers’’ of energy. For thin beams the magnitude of the energy carried by the wave is proportional to the square of the amplitude of the traveling wave. Thus if traveling waves in structures can be sensed in the time domain then the corresponding power flow will be estimated in the time domain. In this paper, a time domain method for the estimation of flexural and extensional waves in beams will be presented. The method uses a spatial array of sensors in conjunction with digital filters to estimate the separate power flow associated with positive and negative traveling flexural and extensional waves. In an example, surface- -mounted PVDF sensors were used in conjunction with a digital filter network to estimate the traveling waves for bandlimited spectra. Experiments where conducted in which the simultaneous flexural and extensional power flow in semi-infinite and finite beams was controlled using the wave vector sensors and surface mounted piezoceramic actuators. These experiments demonstrate the control of beam vibration using power flow-based methods requires fewer actuators and sensors then corresponding modal control techniques. [Work supported by NASA Langley Research Center.]
Sensor location considerations for active noise control in enclosures
Authors: Parkins John W. , Sommerfeldt Scott D.
Published: October 2005

Minimizing the squared pressure at a discrete point(s) is one method of achieving global control in an enclosure, but this strategy will fail when the error sensor(s) lie close to nodal planes of the pressure field. In this case, the secondary modes dominate the pressure measurement, and the active control will create a minimum with little consideration given to the dominant mode. Subsequently, primary mode amplification may result, and the total potential energy in the enclosure will increase. A control based on energy density, on the other hand, can generally sense the dominant mode when the error sensor is close to a pressure field nodal plane, due to its dependence on velocity as well as pressure. Nodal patterns of the energy density field consist of nodal lines and nodal points that lie on the pressure field nodal planes. At these locations, energy density measurements will also be dominated by the secondary modes, and may cause primary mode amplification. Computational results of pressure and energy density fields will be presented which provide insight to optimal error sensor placement for the two aforementioned control methods.
Active control of structural volume velocity using shaped PVDF sensors
Authors: Berry Alain
Published: October 2005

In low frequency, the sound power radiated from planar structures is simply related to the net structural volume velocity. A cost function based on the volume velocity in active control of structural radiation has the advantage of keeping the control simple (one error sensor). The implementation of volume velocity error sensors in feedforward control of flexural beams and plates using shaped PVDF films is presented. For a beam, a single extended strip of prescribed shape is needed, while in the case of a panel, a number of shaped strips related to the number of flexural modes contributing to the volume velocity is required. The sensor obtained is independent of the type and frequency of excitation. A procedure for deriving the appropriate sensor shape, based on analytical or experimental modes, is discussed. The experimental implementation of volume velocity sensors is addressed and results of active control using piezoceramic (PZT) actuators are presented in the case of a simply supported beam, and simply supported or clamped panels. The strategy of minimizing the volume velocity is shown to provide significant acoustic attenuation for structural free-field radiation or transmission problems.
Active control in three-dimensional enclosures using multiple secondary sources and error sensors
Authors: Sommerfeldt Scott D. , Parkins John W.
Published: October 2005

The use of multiple secondary sources and multiple error sensors can significantly improve global attenuation whether one employs a control method based on the squared pressure or energy density. A single source positioned close to a pressure node will be inefficient at exciting the corresponding mode, therefore the secondary modes will dominate the pressure field, and attenuation is unlikely at the related frequency. Increasing the number of secondary sources improves the probability that at least one source will not lie close to a pressure node, thereby mitigating this problem. Problems also arise when error sensors are close to nodes. Adding multiple error sensors increases the probability that the sensors will be able to observe the dominant modes, which will yield improved attenuation. Using a greater number of error sensors than secondary sources will yield a determined control system, with a unique optimal solution. If more sources are used than sensors, an underdetermined control system will result which can be uniquely solved by adding more constraints to the system, such as minimum effort. The performance of the energy density versus squared pressure control methods are compared as they relate to the use of multiple secondary sources and multiple error sensors.
A structural-based acoustic intensity sensor
Authors: Carneal James P. , Gibbs Gary P.
Published: October 2005

A structural-based acoustic intensity (SBAI) sensor has been developed for low-frequency applications. The sensor is comprised of a structural mounted accelerometer and pressure sensor. Local acoustic intensity is calculated by the time average of the product of the velocity and pressure measurements. The ability to use a structural-mounted pressure sensor was confirmed when the phase between a near field and structural-based pressure sensor was determined to be less than 5 deg up to approximately 1500 Hz. Verification of the SBAI sensor proceeded as follows. The structural-based acoustic intensity sensor output was shown to be proportional to the output of a calibrated two microphone intensity sensor from a piston source. Active control of the piston source using the SBAI as an error sensor showed global reduction of radiated acoustic power of approximately 15 dB for several harmonic excitation frequencies. Active control of a complex structure (plate) at various frequencies displayed mixed results. For the (3,2) resonance and a position of (x/Lx=0.85, y/Ly=0.78), global reduction of radiated acoustic power of approximately 9 dB was achieved. However, other tests did not show this type of reduction primarily due to the SBAI sensor acting as a local intensity estimator. It is evident that when applied to a complex structure, the point (or points) at which the intensity measurement is taken must be chosen carefully to obtain a global estimate of the far field radiation.
One-dimensional sound field control by sound pressure and energy-density sensor
Authors: Soditus J. Warner , Tichy Jiri
Published: October 2005

The use of an energy density error quantity for adaptive filtered-x active noise control has been demonstrated to provide advantages over pressure squared control in a one-dimensional experiment [S. D. Sommerfeldt and P. J. Nashif, ‘‘An Adaptive Filtered-X Algorithm for Energy-Based Active Control,’’ J. Acoust. Soc. Am. 96, 300–306 (1994)]. The performance of energy-density-based control was further explored in comparison to pressure-squared control. By using two small microphones, the pressure and energy-density sensor was moved through regions of the upstream and downstream of the secondary source in a one-dimensional low-frequency harmonic sound field. Extensive data were collected for various sensor locations to demonstrate that the pressure-based control strongly depends on sensor location, while the energy-density-based control is more sensor location-independent.
Optimization of multiple piezoelectric actuator locations for active control of sound radiation from plates with arbitrary boundary conditions
Authors: Song Tao , Fuller Chris R.
Published: October 2005

Active control of sound radiation from plates with arbitrary boundary conditions is studied in this paper. An optimization procedure for the locations of piezoelectric actuators is developed. The objective function in the optimization is chosen as the sound power radiated from the plates. The influence of the changes in the support conditions on the eigenproperties of the plate is evaluated. The numerical results show that the control performance with respect to a boundary condition change is dependent upon the excitation frequency for the case where actuators are optimized for a certain boundary condition. However for the case of simply supported boundary condition, it is observed that a change in the translational stiffness will result in degraded control performance only when the translational stiffness is significantly less than infinite (i.e., simply supported). [Work supported by ONR.]
Popular music performance and acoustics in spaces designed primarily as sports halls
Authors: Wrightson Jack
Published: October 2005

Large indoor sports arenas have always served a secondary function as venues for other types of entertainment and as gathering spaces. The reasons include available capacity to serve large crowds, proximity to dense population centers, and the economic needs of the facilities themselves, which cannot afford to operate as a single-season sport franchise. The rise in popularity of the touring ‘‘rock acts’’ is examined, as well as other types of musical presentations, and some of the acoustical planning considerations needed to reasonably successfully accommodate the diverse acoustical needs of various types of entertainment in these halls are discussed. Examples and lessons from recent work will be presented.
Current trends in pop music recording facilities
Authors: Berger Russ
Published: October 2005

The design of current recording facilities is being influenced by several factors. The proliferation of digital signal processing equipment, signal acquisition, and storage media have sparked changes in the way recording studio facilities are used. The mixture of programmed synthesis with acoustic studio performance and the resulting changes in recording technique also provide a significant influence on the design of pop music recording facilities. The impact these changes are having on facility layout, configuration, and function; noise and vibration control; the architectural acoustic environment of studios and control rooms; and the electroacoustic interface will be discussed.
Auditorium characteristics and a cappella music
Authors: Ryan William W.
Published: October 2005

Many authors have presented the typical curves of reverberation time versus volume for various auditorium functions, such as religious, concert hall, opera, etc. Most of the religious curves suggest that the auditorium will be used for instrumental (piano or organ) accompaniment of the singing, whether congregational or choir. This paper presents the preliminary results of a congregational survey to discover what the participants of a cappella music desire. A plot of the reverberation times and volumes for the auditoriums of the participating congregations is also presented. From these preliminary results, it appears that the desire is for a more lively environment, and that the reverberation time should be somewhat higher than that of the ‘‘protestant church’’ curve as presented in Acoustics by Beranek. [Work supported by Ryan & Associates.]
Music critic listening experiment at Dallas’ Meyerson Symphony Center—A progress report
Authors: Lubman David , Siebein Gary W.
Published: October 2005

Professional music critics often pass judgment on concert hall acoustics in their reviews. Because of the wide circulation given to their reviews, music critics may strongly influence public perceptions of the acoustical merits of concert halls. Surprisingly, no systematic study has been reported comparing critics’ acoustical judgments with those of musicians and of ordinary listeners in the same hall, despite its potential value to both acoustical and critic communities. For this reason, the ASA launched a cooperative effort with the Music Critics Association of North America (MCANA). Its first step was an informal experiment at Dallas’ Meyerson Symphony Center during MCANA’s June 1994 meeting. The critics’ meeting provided a potential opportunity to obtain acoustical assessments from a significant number of critics attending symphony concerts in the same hall. Standard survey forms and instructions were placed in the registration packets of about 75 music critics at check-in, and were distributed randomly to about 1000 volunteers attending concerts on two consecutive nights. Disappointingly, of 138 survey forms returned so far, only about 10 are from critics. This could improve by presentation time. Survey results and lessons learned will be reported. [Work supported with private donor and TCAA Technical Initiative Funds.]
Computer-based system for reverberation room design
Authors: Castañeda Eduardo Méndez
Published: October 2005

A simple computer system named SIDIART developed in QBASIC language is proposed, which permits the manipulation of a database of materials and their respective absorption coefficients, making their election easy. Knowing the volume and the areas, if the user wishes, the system suggests the optimal 500-Hz reverberation time (RT) from one of ten optional types of rooms. The Sabine or Eyring equations, and relative humidity between 10% to 90% for the calculus of air absorption, can be chosen. Given a number of surfaces, its areas and a group of candidate materials for each surface, the system reports the ratio between the optimal and calculated RT for each possibility and for each frequency. The system has the option of searching and reporting only the combination that offers deviations from the optimal RT defined by the user as acceptable, if that combination exists. The algorithm of search is based on the traditional way to solve the problem without a computer. The advantage of the system takes root in the velocity of calculus provided by the computer; however, the final result of the project depends upon the designer.
Sound isolation at low frequencies, only an extension of the traditional frequency range?
Authors: Kihlman Tor
Published: October 2005

The expression ‘‘low frequencies’’ implies a frequency range where the rooms’ dimensions are comparable with the wavelength of sound in the rooms. In this range (below 150 Hz for normal room sizes) the sound field in the rooms consists of a few modes only, and changing of parameters that are not connected to the partition influences the sound insulation. This is not a problem of a correct measurement procedure. It is far more the description of the sound insulation itself in which problems arise. The measured or calculated sound insulation is only valid for the specific case under consideration. Numerical studies of the sound insulation of a flexible partition (characterized by a limp mass) between two rooms show that the dimensions of the sending and receiving rooms have the most important influence. This has to be taken into account when extending the frequency range in the standards concerning sound insulation which is necessary since neighbors low frequency stereo equipment, as well as the all present traffic noise from outdoors, demands a sufficient sound insulation even below 100 Hz.
Sound isolation—A review
Authors: Rivas Ilhuicamina Servín
Published: October 2005

When treating the property of sound attenuation provided by partitions, different ways are found for its specification. In this paper most common descriptors employed to specify airborne and structureborne sound insulation are summarized and reviewed. Advantages and disadvantages of single-number specifications for sound insulation are commented upon. On the other hand, emphasis is made as to the importance and convenience of field measurements of sound insulation, even though they are time-consuming. Also included is a brief description of the situation prevailing in some countries, Mexico among them, regarding minimum sound insulation requirements between dwellings. Finally, reference is made to the use of acoustic intensity techniques for the measurement of sound insulation and the need of standardization on this type of evaluation.