bluuebunny/crossref_metadata_2025 huggingface dataset:
Use of data reduction in computer music
Authors: Beauchamp James W.
Published: January 2006

Two competing methods of digital sound production are “instant replay” and “synthesis by algorithm.&” If fidelity to an original sound were the paramount consideration, replay would be the more desireable method. However, replay requires massive amounts of memory and is inflexible with respect to contextual modification, an important concept for composers. On the other hand, algorithmic synthesis methods generally are designed to minimize memory requirements and employ a few, well-chosen, perceptually significant parameters which are varied in an ad hoc fashion. Data reduction, where synthesis algorithm parameters are automatically extracted from the original sound, offers a link between instant replay and algorithmic synthesis. It is particularly useful if the extracted parameters are perceptually meaningful and therefore can be intelligently modified by the composer. Various methods of data reduction will be discussed together with their associated synthesis algorithms (including constant waveform, additive, filter, nonlinear/filter, and waveform interpolation synthesis), and sound examples will be played.
Modification of musical sounds by means of the phase vocoder
Authors: Dolson Mark
Published: January 2006

The usefulness of the phase vocoder for performing time-scale modification of speech signals is well established. However, the application of the phase vocoder to musical signals is far less common. Musical applications of the phase vocoder differ from speech applications in several important respects: (1) Musical sampling rates are much higher; hence, minimization of computation and storage requirements is far more important. (2) Musical fidelity requirements are also much higher; this counteracts attempts to minimize computation and storage. (3) Musical applications often require analysis of individual harmonics; in this case, filter spacing must match harmonic spacing. (4) Pitch translation (as opposed to time-scale modification) requires spectral envelope estimation to sound convincing. (5) Musical signals may be polyphonic. (6) Lastly, musical applications frequently seek to distort sounds in an interesting way; hence, they often involve extremes of time and pitch scaling, selective modification of analysis data, and novel cross-syntheses and timbral interpolations. Examples will be presented from a phase vocoder package designed explicitly for musical applications.
High-frequency acoustics in the Arctic
Authors: Francois Robert E.
Published: January 2006

Compared with other oceans, acoustic propagation in the Arctic is generally limited in range by upward refraction, ice surface interaction, and the increased absorption loss at low water temperatures. At shallow depths beneath the under-ice surface, the water column is very near its freezing point during most of the year. Considerable salinity variation occurs in the freezing season because of brine displacement as ice is formed and in the melt season because of the layering of the nearly fresh ice melt water over the sea water. This results in a positive density gradient with a rather sharp increase at 20–50-m depth, where the older Arctic water is encountered. A dense biological population, analogous to the deep scattering layer seen in the open ocean, is observed at this pycnocline. This thin layer, has high volume reverberation strength compared to values observed in other oceans. A study of these layers sometimes reveals the presence of internal waves. When propagation takes place through this layer, anomalously high absorption attributed to scattering is observed. At 60–300 kHz, we have observed apparent increases in absorption of from 10 to 20 dB/km. The presence of the ice cover decouples the wind from the water mass, resulting in greatly reduced boundary layer mixing. This allows the many discrete layers formed by the natural processes noted above to persist over large areas for long times. As a consequence large acoustic fluctuations often occur for small vertical displacements of source or receiver. [Work support by NAVSEA.]
Site dependence of low-frequency Arctic ambient noise and signal attenuation
Authors: Diachok Orest I. , Wales Stephen C.
Published: January 2006

Our understanding of two significant, ice-related, low-frequency (below 1 kHz) underwater acoustic phenomena, characteristic of both Arctic and sub-Arctic environments, is briefly reviewed. Measurements suggest that ocean wave-generated noise at the periphery of the ice-covered Arctic Ocean is generally exceptionally high when the fetch is large, and can have very long range effects, when propagated signals are not subject to significant surface scattering and volume absorption losses. Surface scattering due to sea-ice ridges, which may be modeled as randomly distributed elliptical half-cylinders, dominates under-ice propagation at frequencies above about 20 Hz. Ray theoretical calculations incorporating this model are consistent with observed site-dependent transmission loss differences over a limited frequency range (40–1000 Hz). Results of new lower frequency wave theoretical calculations, which incorporate this model and a suitable geoacoustic structure of the ocean bottom, will be presented and compared with data. Initial concepts for extension of this work to three dimensions (including randomly oriented ridges) to enable more refined predictions of scattering/transmission losses and, in particular, spatial coherence losses will be outlined.
Measurements of the vertical fields of acoustic signals and normal mode propagation in the Arctic Ocean
Authors: Yang T. C.
Published: January 2006

The vertical sound fields of cw signals and shots propagating in the ice-covered Arctic Ocean were measured during the FRAM IV experiments in April 1982. The measurements were made with a vertical array consisting of 28 hydrophones extending to a depth of 960 m. This experiment marked the first successful deployment of a large aperture vertical array in the Arctic environment. This paper will present the results and interpretations of the vertical array data. The lowest order normal mode amplitudes and phases were measured from the shot signals received simultaneously on the 28 hydrophones of the vertical array. The measured mode amplitudes arc compared with the computed normal mode and used as the basis for spatial filtering of normal modes from the cw signals. Intensity distributions of the cw signals versus depth are presented for several frequencies below 100 Hz. Spatial coherence of the cw signals are studied and compared with that of the ambient noise. The arrival angles of the lowest order normal modes are measured from the cw signals using the mode filtering technique. The angular arrival pattern of the cw signals are plotted using conventional beamforming. Temporal behavior of the cw signals will be shown; phase tracking beamforming is used to enhance the vertical array gain.
The finite element method in acoustics
Authors: Geddes Earl
Published: January 2006

From the variational field equations for problems in acoustics one can derive the finite element approach to their solution. From this approach the application of a standard finite element method (FEM) program to a problem in room acoustics was performed. The role of the acoustic variables, the material constants, boundary conditions, etc., as required for the FEM program was examined. The need for complex arithmetic in the analysis of a nonconservative system and in the modeling of real sources was shown. The FEM eigenvectors and eigenvalues were then used to assemble the Green's function in the form of a data base. This Green's function was then to be used to examine the acoustic behavior of the room at low frequencies, i.e., below the Schroeder frequency.
An application of SEA to helicopter structure borne noise
Authors: Hawkings D. L.
Published: January 2006

The paper describes an exploratory application of SEA to helicopter structure borne cabin noise. The work was focused on a fall-scale center cabin test section of advanced structural design, rather than a complete helicopter fuselage. Several theoretical models were developed, differing in the complexity of the structural representation adopted. The main difficulties encountered in the modeling process arose from the composite sandwich construction of the wall panels, and in particular expressing the coupling loss factors from the beamlike frame elements to these panels. Numerical predictions based on these theories were made of the expected vibration levels around the structure, and the acoustic pressure within the cabin. These predictions were compared with measured results from the trial structure. Good overall agreement was found with the more refined model, although the cruder model still gave acceptable prediction of noise levels across the frequency range considered (250–8000 Hz). It is concluded that SEA is potentially a valuable tool for predicting helicopter cabin noise. [Work supported by U. K. MOD.]
Active control of noise in enclosed spaces
Authors: Warnaka Glenn E. , Zalas John M. , Tichy Jiri , Poole Lynn A.
Published: January 2006

Active methods for controlling noise have been demonstrated to be feasible. As they are compact in volume and low in weight, active controls seem very promsing for controlling low-frequency noise in enclosed spaces, particularly in transportation vehicles. Nevertheless, certain difficulties arise due to the modal structure developed within the enclosed spaces, because the cancellation sources must produce an exact match of the unwanted noise field in order to cancel it. Similarly, the nonuniform nature of the noise field poses problems in detection of the noise. This paper discusses the progress which has already been made in controlling interior noise and shows that there are several approaches that may be adopted. The methods depend on whether the noise originates within the space or from sources outside the space. If the sound enters the space by penetration from the outside, then the method of penetration can be important in determining the location of the cancelling sources and the detecting microphones. The technical potential and the latitude of application indicate that there is considerable promise for active control of noise in enclosed spaces.
Comparison of procedures for determination of acoustic nonlinearity of some inhomogeneous materials
Authors: Bjørnø L.
Published: January 2006

The widespread use of ultrasonic energy in medical diagnosis and therapy and the recent research showing the existence of nonlinear acoustic effects in inhomogeneous materials, like biological media, have led to a strongly increased interest in investigations aiming at providing enough information for an evaluation of the second order nonlinearity ratio B/A as a tissue characterizing parameter on a par with attenuation, impedance, etc. The several techniques used for the determination of B/A roughly fall into two main groups, finite-amplitude methods and thermo-dynamic-acoustic methods. The B/A values determined using various techniques of the two main groups are strongly scattered, showing a dependence on the procedure used, which makes it difficult to draw any valid conclusions. The paper discusses advantages and disadvantages of the techniques used hitherto, aiming at an explanation of the reasons for the deviations between the present B/A results for biological media. The necessary accuracy in the determination and the reliability in the exploitation of B/A as a tissue characterizing parameter for clinical use are emphasized.
Ultrasonic study of oil/water emulsions stabilized with nonionic block copolymers
Authors: Gultepe M. A. Barrett , Gultepe M. E. , McCarthy J. L. , Yeager E. B.
Published: January 2006

An increment in sound velocity at 2 MHz has been observed as the emulsion globule size decreases with time in a water, toluene, and polyoxethylene (POE)—polyoxypropylene (POP) block copolymer system. The observed effect depends on the copolymer concentration and is largest for the block copolymer composed of 50% POE–50% POP moieties. A general acoustical treatment taking into account viscous loss in the emulsions globules as well as surrounding liquid and heat conduction across the interface fails to explain the observed velocity increment. Aqueous solutions of the above copolymer shows association effects as evidenced by a break in the sound velocity versus concentration curve at approximately 15 g/l. [Work supported by ONR.]
Acoustic signals having both positive and negative nonlinearity
Authors: Cramer M. S. , Kluwick A.
Published: January 2006

The quantity Γ≡(1/ρ)(∂ρa/∂ρ) frequently occurs in the nonlinear acoustics of both gases and liquids. Here ρ is the density, a the sound speed, and the derivative is taken at constant entropy. It is well known that when Γ > 0 (positive nonlinearity), pulses and periodic wave trains steepen forward to form compression shocks. When Γ < 0 (negative nonlinearity), the steepening is backward and expansion shocks form. When the signal amplitude is sufficiently large compared to Γ, the local value of Γ may vary from point to point in the wave and the subsequent behavior is quite different than the cases having either positive or negative nonlinearity everywhere. This phenomena can occur for single phase Navier-Stokes fluids and the present paper will provide a description of the propagation in this latter case. Results of interest include shock splitting, shock waves having sonic conditions either upstream or downstream of the shock, and collisions between compression and expansion shocks. Under certain conditions it was found that the ultimate decay of a mono-signed pulse is proportional to the negative one-third power of the propagation time rather than the classical negative one-half power.
Dark-field magneto-elastic nondestructive testing technique
Authors: Chamuel Jacques R.
Published: January 2006

In the field of nondestructive testing, great advantages can be obtained if a dark-field technique can be achieved. A magneto-elastic technique generating sound waves only on the presence of a defect and from the defect is described (no defect—no sound waves in the material under test). The magneto-elastic technique generates three-dimensional information about the defect location, dimensions, and characteristics, unlike conventional ultrasonic NDE techniques where the defect is used as a scatterer of incident ultrasonic waves. Eddy currents are generated in the material under test with a current carrying conductor. A magnet is placed such that its magnetic field is parallel to the current carrying conductor. Elastic waves are generated in the conductive material under test only at the crack edges from the interaction of the magnet with transverse eddy currents flowing around the defect. The generated elastic waves are detected with an ultrasonic receiver transducer coupled to the material. An array of receivers may be used to triangulate on the defect and also to characterize the properties of the defect from the 3-D information. Conventional EMAT (electromagnetic acoustic transducers) are used to generate elastic waves in the materials even when no defects are present; the magneto-elastic technique depends on the presence of a defect to create the elastic waves. Preliminary experimental results are presented together with other potential application of the new concept.
Self-scanning-line electro-elastic NDE technique
Authors: Chamuel Jacques R.
Published: January 2006

A contactless self-scanning-line electro-elastic nondestructive testing technique is described capable of determining in real time the presence, location, number, and size of defects present along a linear path on the surface of a conductive material under test. In one embodiment, the NDE probe consists of a magnetostrictive delay line placed in close proximity with the surface to be inspected. A current carrying wire placed adjacent and parallel to the delay line is used to induce eddy currents in the material. The presence of a crack creates transverse eddy currents interacting locally with the magnetostrictive delay line launching elastic waves in the delay line. The elastic waves are generated only at the crack edge locations along the delay line. Spatial information regarding crack edge locations is preserved in the time information. An ultrasonic receiver coupled to the delay line is used to detect the elastic waves traveling in the delay line. The self-scanning-line electro-elastic NDE probe is not susceptible to the presence of elastic waves propagating in the material under test. The new concept is applicable to flat and curved surfaces including pipe inspection. Conventional eddy current techniques are not self scanning along a linear path giving only an integrated response, indicating only the presence and not the location and number of defects. Application of the concept to automation will be discussed.
EPDM rubber as an underwater acoustic window
Authors: Thompson Corley M. , Capps Rodger , Lizzi Michael J.
Published: January 2006

The ethylene-propylene-terpolymer (EPDM) class of elastomers has been widely suggested but rarely used in underwater acoustic devices. Recommendations for its use usually result from a recognition of its low acoustic losses and good match of sound speed and density with seawater. The hesitations about using EPDM's result from unknowns in its engineering properties of bondability and tear strength. The reported marginal bondability and tear strength were investigated as a function of EPDM formulation. Specifically, the effects of filler type and amount, sulfur or peroxide cure type and loading, and polymer parameters on both engineering and acoustic properties will be shown. With care in designing the EPDM formulation, this material can have acoustic properties (both rhoc and attenuation) as good as natural rubber and environmental resistance superior to neoprenes. EPDM's compatability with castor oil is the best of all common classes of elastomers. The results of many tests confirm that while EPDM's bondability and tear strength are not equal to neoprenes, they are nevertheless adequate for most applications.
A speaker recognizability test for communications systems
Authors: Papamichalis Panos , Doddington George
Published: January 2006

A Speaker Recognizability Test (SRT) has been designed which tries to establish how well a given communications system preserves a speaker's identity. Contrary to previous efforts, no attempt is made to identify the cues used by listeners for speaker recognition. Instead, listeners are asked directly to identify a speaker who says an utterance. The test is constructed as follows: Several sentences are collected from five male and five female speakers. One sentence from each speaker is used as reference. The listening team consists of ten listeners. Each listener is presented 20 different sentences and is asked to identify the speaker of each one of them by comparing it with the ten reference sentences. Among the issues considered in the design of the test is the choice of speakers, the use of reference sentences from the same or different sessions of data collection, and the use of processed or unprocessed speech for reference.
Does a Bernoulli force cause velar double bursts?
Authors: Kingston John
Published: January 2006

Multiple intensity peaks, “double bursts,” are common in the release of velar stops. Keating, Westbury, and Stevens [J. Acoust. Soc. Am. Suppl. 1 67, S93 (1980)] suggest that shortly after the velar release, a rate of air flow arises between the tongue dorsum and the soft palate high enough to pull them back together by a Bernoulli force like that which draws the vocal folds together. The release of this second contact produces a second burst. If a high rate of air flow causes this second, brief closure, then double bursts should occur more often after /k/ than /g/, since intraoral air pressure during closure is higher for the voiceless stop than the voiced one. Intensity measurements of the initial portions of the bursts of [kh,sk,g,th,st,d] before [i,ʌ,u] show vowel quality determines burst intensity for velars—least intense before [u], most before [i], with [ʌ] inducing weak to moderate bursts—more than the voiced/voiceless or aspirated/unaspirated contrasts do. Alveolar releases vary much less across vowels, but peak intensity is reached approximately 10 ms later after [th] than [d] or the [t] which follows [s]. Evidently, the point of contact with the soft palate or the movement to the following vowel mechanically determines burst intensity for velars, while for alveolars aero-dynamic factors are more potent.
The violin octet
Authors: Hutchins Carleen M.
Published: January 2006

Eight new violin family instruments in graduated sizes project the tone quality of the violin into seven other tone ranges, representing the first time a consistent theory of acoustics has been applied to a family of musical instruments. Undertaken in 1956 at the instigation of Henry Brant, they resemble instruments constructed in Europe during the 16th and 17th centuries. Using early violin tests, F. A. Saunders, John C. Schelleng, and C. M. Hutchins ferreted out a controlling characteristic of the violin: namely the two main resonances within a semitone of the two open middle strings. These two resonances are differently placed in viola, cello, and bass. Schelleng and Hutchins developed a mathematical scaling theory and Hutchins spent nearly 10 years adapting and constructing instruments according to these parameters. The first concert was at Harvard in 1963, a memorial to F. A. Saunders; a second in 1967 at the New York YMHA. There are over 50 compositions and arrangements for the OCTET, which has traveled thousands of miles for concerts and lecture demonstrations—one set permanently in England, another in Stockholm, with a composer contest and concert for the “Swedish Musical Acoustics Conference 1983.” Violin makers in the USA, Europe, and Australia are constructing the instruments which are described in “THE VIOLIN OCTET” available from the Catgut Acoustical Society. Personal work time privately donated; machinery and acoustical equipment funded by: John Simon Guggenheim Memorial Foundation, Martha Baird Rockefeller Fund for Music, American Philosophical Society, Arnold Hoffman Foundation, George MacDonald Foundation, Estates of Virginia Apgar, Helen Rice, Eunice Wheeler, and the Harriett M. Bartlett Fund of the Catgut Acoustical Society, Inc.
An inflatable guitar
Authors: White Timothy P.
Published: January 2006

It is readily demonstrated that blocking the soundhole of a steel-string acoustic guitar does not significantly affect the instrument's tone. Wavelength considerations indicate that the high frequencies to which our ears are most sensitive radiate directly from the guitar's soundboard. It is common knowledge among luthiers that, in determining the tone of an instrument, the stiffness and damping properties of the guitar's soundboard are of paramount concern. In this study, the traditional wooden architecture of the guitar's soundboard, typically a spruce veneer reinforced with variable wooden bracing, is replaced on a commercial production instrument with a graphite-epoxy “grill” supporting a thin synthetic polymer membrane under tension. In another instrument, the guitar's traditional wooden soundbox is replaced altogether by a graphite-epoxy soundboard grill backed by an inflatable bladder in the shape of a normal guitar body. Both instruments are demonstrated, and techniques for creating aligned-fiber graphite instrument components with room-temperature “time-set” rather than “thermo-set” epoxy resins are discussed.
Audiograms for large cetaceans: A proposed method for field studies
Authors: Ridgway Sam H. , Carder Donald A.
Published: January 2006

In the great majority of cetaceans, the sense of hearing has not been studied. There are no audiograms on any of the great whales. To us, the ABR (Averaged Brainstem Response) technique seems the only way to get such data on the larger species. We have used the ABR in smaller cetaceans such as Delphinus, Tursiops, and Delphinapterus to record audiometric data. Fine wires about the size of a human hair are inserted into the head through a hypodermic needle. Bare tips of the insulated wire over the vertex and mastoid region pick up electrical potentials (electroencephalogram or EEG). The EEG is amplified, filtered, and tape recorded during periods when sound pulses are presented through a hydrophone. We observe the ABR by synchronous averaging of the EEG for 10 to 25 ms after each pulse. The sound intensity and frequency are varied to determine the threshold at each frequency, thereby producing an audiogram for the animal. Audiometric data would be helpful in assessing the impact of human generated acoustic events on whales. We have developed a portable system that we hope will enable us to make measurements on whales trapped in fishermen's nets, cetaceans stranded in tidal basins, or animals that beach themselves.
Sound and vibration levels in a ringed seal lair from seismic profiling on the ice in the Beaufort Sea
Authors: Holliday D. V. , Cummings W. C. , Bonnett D. E.
Published: January 2006

The ringed seal (Phoca hispida) habitat on Alaska's North Slope is subjected to intensive local seismic exploration. During the winter and spring months, this seal constructs and maintains lairs for resting and birthing on the ice surface under a snow cover. Measurements of airborne sound and vibration levels in a lair and underwater sound levels beneath the lair are reported as a function of range to a seismic sound source (VibroseisR) operating on the ice near Reindeer Island, north of Prudhoe Bay, AK. Propagation losses for underwater sound in the band between 10 and 70 Hz were from 10 to 20 dB per doubling of range. Vibration energy losses range between 20 and 40 dB per range doubling for the same frequencies. Losses increased with increasing frequency. [Work sponsored by NOAAs National Ocean Survey, Office of Oceanography and Marine Services.]
Theoretical and experimental study of porous absorbing material
Authors: Berry J.-L. , Nicholas J. , Malric B.
Published: January 2006

Porous ceramic has raised considerable interest given the great number of applications of its absorbing and insulating properties, both indoors and outdoors. Numerous experiments with a Kundt tube enable us to evaluate the sound absorbing characteristics of ceramic samples with variable flow resistivities (5–100 cgs), thickness (20–90 mm) and porosities (60%–80%). The most important factors were shown to be flow resistivity and thickness. Subsequently, a theoretical analysis was carried out in terms of normal incidence absorption coefficient using the following impedance models: Delany and Bazley, Beranek, Cremer, and more recently Attenborough. Material thickness and the presence of rear air space (or other material) are also taken into account in order to simulate more accurately practical situations. It was found that Cremer's approach adapted empirically to ceramics gives the best prediction. Thanks to this study, the optimal flow resistivity and thickness may be determined to obtain the best cost/performance ratio.
Sound propagation in Hell with partially locked normal component
Authors: Baker Steve , Rudnick Daniel , Rudnick Isadore
Published: January 2006

We consider the consequences of introducing a flow resistance, R, into the superfluid equations of HeII. We find: (1) The dimensionless quantity that characterizes flow resistance in HeII is R/ρπω≡r. (2) In a superleak there is a continuous transition from first to fourth sound and from second sound to a diffusive thermomechanical (TM) wave as r increases from 0 to ∞. (3) In a pressure released system (such as a shallow puddle of HeII on a flat substrate) there is a continuous transition from second sound to a TM surface wave and from gravity wave to a diffusive TM surface wave as r increases from 0 to ∞. (4) For each system the attenuation per wavelength for the propagating mode is a maximum near r = 1, decreasing monotonically as r→O, ∞. [Work supported by ONR.]
Magnetic recording, past and present
Authors: Camras Marvin
Published: January 2006

Why is the old complaint true: “The ancients have stolen our inventions?” There are good reasons: It is easy to propose something, difficult to make it work. Excellent ideas are often discarded when a trifling mistake gives negative results; later a luckier discoverer gets the credit. Case histories in magnetic recording illustrate these points. Metal particle tape was proposed 95 years ago. High frequency bias had its roots in the Marconi-Muirhead wireless detector of 1902. Digital recording goes back at least to Morse's telegraph of 1840. The rotating-head principle used for video recorders dates back about three decades before it became practical. Sendust used in the latest recording heads was invented in 1936. Magnetic disks, the important memory of personal computers, were demonstrated in about 1905. Soundstriping for motion picture film goes back to the early 1920s. Personal experiences of failures and successes are related.
Fifty years of stereo phonographs: A capsule history
Authors: Anderson C. Roger
Published: January 2006

From the 45° disk recording system shown in the 1933 benchmark stereo work of Blumlein to the present day generations of pickups with ultrasonic capabilities is a long evolution of analytic and engineering effort. This paper is a recognition of advances along the way, such as the theory of tracing distortion, stylus contours, transducer principles, light and stiff moving systems, tracking requirements, and the design insight that led to contemporary pickups which perform superbly at 1-g stylus force.
Sound intensity errors in a reactive field
Authors: Nicolas Jean , Lemire Gilles
Published: January 2006

During the past years many in situ intensity measurements have been conducted by the authors looking ahead for a better understanding of the phenomena involved when quantifying sound power of a particular source in the presence of external perturbating noise. It became evident that the limitations depend on the following question: is the field progressive or reactive? A general formulation is developed for both types of field. In a reactive field, the fact that the instantaneous phase variation of the pressure is different from that obtained for a progressive field is taken into account. This leads to enormous uncertainties that have no common magnitude with errors associated with progressive field. In order to determine these uncertainties when measuring sound power, a model with two or more point sources creating various degrees of reaction on the measurement envelope is used. It shows the action and interaction of the main parameters (power ratios of the sources, distance, number, and spread of measurement points). Interestingly enough it is shown that an increase in the number of points does not reduce the uncertainty in the case of a highly reactive field.
Acoustic intensity measurements in the reflective environment
Authors: Mielnicka-Pate A.
Published: January 2006

The two-microphone intensity method was used to investigate the intensity distribution and radiated power of several sources. The sources used were two open pipes of 30-mm and 95-mm diameter and an aluminum circular plate of 100-mm diameter. The pipes and plate were mounted with baffles. The investigations were confined to the low-frequency range of plane-wave propagation inside a pipe. The experiment was performed in the nearfield and farfield of the sources in the reflective environment. The theoretical and experimental results will be discussed for both intensity and radiated power. The influence of the source impedance will be analyzed for the intensity measurements, particularly at points located close to the source. The intensity error that can occur due to the partial absorption of the reflected waves by the source will be discussed.
Probe measurement of eardrum SPL in occluded real ears
Authors: Gilman Samuel , Dirks Donald D. , Hanson David G.
Published: January 2006

Previously, we reported results from simulator studies of techniques for remote measurement of eardrum SPL in hearing aid occluded ears. These data provided means for calibrating remote-measuring probes which included the effects of ear canal length and impedance at the eardrum. The present paper presents data with respect to the accuracy of probe/earmold measurements of eardrum SPL in real ears. Measurements were made on fresh cadaver ears in which the eardrum SPL was independently measured with a micro-miniature microphone placed adjacent to the eardrum. Simulator-derived corrections obtained from the previous study were applied to the probe/earmold measuring system to compensate for ear canal length and assumed eardrum impedance. Results from several ears showed the wide variation associated with individual ear measurements and corrected probe/earmold errors of less than ± 4 dB for frequencies up to 6.0 kHz.
Analog filtering of the middle latency auditory evoked response
Authors: McPherson David L. , Montgomery Lance
Published: January 2006

There has been confusion on the origin of wave P0 of the middle latency auditory evoked response (MLR). It has been suggested wave P0 is a residual of the effects of restrictive filtering of wave V of the auditory brainstem response (ABR). Twelve subjects were used to study the effects of analog filtering on wave V of the ABR, and waves N0 through Nc. Monaural clicks were presented at 70 dBnHL. Two series of 1024 averages were collected for a 100-ms sample for nine filter conditions. High-pass filtering caused greater distortion of the waveforms than did changes in low-pass filtering. When high-pass filtering was at or above 150 Hz, P0 could not be consistently identified. It was observed that filtering of both the MLR and ABR had differential effects on ipsilateral versus contralateral recordings. The later MLR waveforms showed greater variability than either the earlier MLR waves, or the ABR wave V for changes in high-pass filtering, especially above 150 Hz. This study is in agreement with the concept of simultaneous recording of the ABR and MLR reported by Scherg [M. Scherg, Electroenceph. Clin. Neuro. 54, 339–341 (1982)] although that study did not directly address the question of filtering effects.
High-pass masker slopes and the BAER
Authors: Hecox Kurt E. , Kilsdonk Tom , Malischke Mary
Published: January 2006

The use of analog filters is a nearly universal practice in masking studies of auditory evoked potentials. Yet there is little systematic information on the impact of certain aspects of the filter's properties. This study reports the results of varying masking noise cutoff frequency (8, 4, 2, 1, and 0.5 kHz) and slope (96, 72, 48, 24, 12, and 6 dB/oct), on the response elicited by a 60-dB HL click. The shallower the filter slope, the longer the latency and smaller the amplitude of the response; the magnitude of this effect interacts with cutoff frequency. There is little evidence of differences between 48 dB/oct and 96 dB/oct slopes, and the maximum change was between 24 and 12 dB/oct for most cutoff frequencies. Since acoustic distortion problems increase with increasing roll-off steepness, it is suggested that 48 dB/oct is an optimum value for brainstem auditory evoked response masking studies. The age dependence of these principles will be shown for newborns. In general, all of the observed latency shifts are consistent with known traveling wave mechanics.
Hemispheric difference in evoked potentials to spatial sound field stimuli
Authors: Ando Y. , Hosaka I.
Published: January 2006

In order to produce different subjective spatial impressions, two sound field stimuli were alternately presented with a low and a high magnitude of interaural cross-correlation (IACC). (Subjective diffuseness is the percept for sound fields with the low IACC.) Auditory evoked potentials (AEP) from both temporal areas (T3 and T4) of five normal subjects for each stimulus were recorded. When a bandlimited white noise was presented as a source signal, amplitudes of AEPs over the right cerebral hemisphere were much greater than those over the left, as similar to previous results by several investigators. However, this tendency was only weak, even if a vowel (a) was presented. Furthermore, it is generally found with the two sound signals that (i) for the sound fields with high IACCS, amplitudes of AEPs over the right hemisphere were always greater than those over the left, and that (ii) time differences between the first and the second maxima of AEPs over the right hemisphere were clearly longer than those over the left to the high IACC fields. From these findings, the right cerebral hemisphere is considered to be dominantly operative for the auditory spatial sensation.
Identification and discrimination of visually presented synthetic consonant-vowel stimuli
Authors: Walden Brian E. , Montgomery Allen A.
Published: January 2006

Two-dimensional vector-based dynamic images representing the consonant-vowel syllables /ba/, /va/, and /wa/ were produced on an intelligent graphic system. The starting configuration and steady-state vowel were identical for these three stimuli. Visual presentation to subjects revealed these stimuli to be highly identifiable representations of the three consonants. Six intermediate stimuli were also generated between each of the three possible pairs of consonants and represented equal transitional steps between the exemplars. The three resulting eight-item continua were presented for identification and discrimination (ABX paradigm). The data revealed that the slope of the identification function in the region of the phoneme boundary and the location of the 50% point differed for the three continua. Identification and discrimination functions, as well as reproductions of stimulus frames, will be presented to illustrate the phoneme boundaries and most distinctive phonemic cues.
Comparison of three performance measures obtained with perceptually balanced vowel continua
Authors: Espinoza-Varas B.
Published: January 2006

The auditory representation of formant frequency appears to be consistent with a scale of critical-band units: the bark scale [Schroeder et al., J. Acoust. Soc. Am. 66, 1647–1652 (1979)]. This paper compares measures of identification, discrimination, and perceptual distance obtained in vowel continua divided into “perceptually equal” steps; i.e., equal steps along the bark scale. The continua ranged from either /u/ to /i/ or from /ɔ/ to /ɛ/. The 400-ms, steady-state vowels were generated using a three-formant parallel synthesizer. Within each continua, the frequencies of F1 and F3 were fixed, while the frequency difference F2 − F1 (or F3 − F2) varied from ≈0.5 to ≈7.0 bark in steps of ≈0.5 bark. The fundamental frequency was 120 Hz. Listeners were asked to identify each of the stimuli in the continua and to make a “vowel-goodness” rating. The results are discussed in the context of the question of whether the bark scale is consistent with all three performance measures [Supported by AHFMR and NSERC.]
Precategorical Acoustic Store (PAS): Is there an auditory sensory memory?
Authors: Williams Diane
Published: January 2006

The existence of an auditory sensory memory called Precategorical Acoustic Store (PAS) was proposed by Crowder and Morton [Percept. Psychophys. 5, 365–373 (1969)] to account for both the modality and suffix effects obtained in list-learning experiments. While PAS provides a good explanation of these effects, the memory has little independent empirical support. PAS was directly investigated in two experiments. In the first experiment, the subject heard a subliminally presented word. The subject's task was to determine whether the two words were the same.In the second experiment, the subjects heard four words presented simultaneously at normal intensity, and later read a word. The subjects' task was to determine whether the word he or she read was one of the words that were heard. If the auditory stimulus was represented in PAS, then performance should be better if the visually presented probe appears before PAS has appreciably decayed, and performance should decrease with increasing probe lags. In both experiments, discriminability performance was roughly constant with increasing probe lags. This failure to support the proposed PAS is discussed.
The pursuit of invariance in speech signals
Authors: Lisker Leigh
Published: January 2006

We may imagine a modified Muller-Lyer illusion in which two lines of unequal length are perceived as equal in different contexts. Is this analogous to the common situation in speech, where two distinct signals, e.g., deep and duke, are judged “equal” with respect to their /d/'s? In the visual case the term “illusion” is susceptible of precise definition, allowing us to speak of a mismatch between stimulus and percept. In the speech situation we are confronted with problems that make it very difficult to pose, let alone answer the question as to whether the /d/ common to deep and duke reflects a perceptual invariance that is illusory. Identity of spelling is only weak evidence for perceptual constancy, since spelling is more akin to phonological than phonetic representation, and at least some phonemes comprise phonetically distinct “sounds.” An example is provided by the voiceless stops of English. It can be shown that acoustically identical segments contribute to bdg in some contexts and to ptk in others. Another example involves /wy/, which may be produced by the same vocal tract configuration in different contexts. Even if some phonemes are marked by acoustic invariants, not all of them are. [Work supported by NICHD.]
Statistical energy analysis for beam networks in building structures
Authors: Sablik M. J. , Beissner R. E. , Silvus H. S. , Miller M. L.
Published: January 2006

The statistical energy analysis method is applied to a beam network in a building structure. A method of taking structural resonances into account is introduced into the analysis. Coupling loss factors are computed using transmission coefficients presented in earlier work [M. J. Sablik, J. Acoust. Soc. Am. 72, 1285 (1982); L. Cremer, M. Heckl, and E. E. Ungar, Structural Borne Sound (Springer-Verlag, New York, 1973)]. Transfer functions computed from the model beam network are compared over a wide frequency range to experimentally measured transfer functions from an equivalent beam network in an existing building structure in which one of the beams is vibrationally excited. Our results emphasize that unless resonance effects are taken into account, the predictions of statistical energy analysis will not display the fine details found in the frequency dependence of transfer functions for real building structures.
Helicopter airframe vibration transmission modeling using statistical energy analysis (SEA)
Authors: Yoerkie Charles A.
Published: January 2006

Predicting helicopter cabin noise is a complex issue. It involves sources which are both acoustic/aerodynamic and vibratory in nature as well as a compact, highly interconnected airframe structure. Energy transmits to the cabin via both airframe structureborne and airborne paths. Since a large number of resonant structural and acoustic modes are excited, a statistical approach, which treats the response of the modes collectively in groups, is desirable. The method described for modeling the helicopter airframe and cabin interior is Statistical Energy Analysis (SEA). An SEA model, funded by a NASA Langley Research Center contract, is presented for the Sikorsky S-76 (a4-13 passenger commercial helicopter). This overall source-to-receiver modeling approach allows for evaluation of various noise control measures in the conceptual design stage. Expressions have been developed for SEA parameters and the coupling loss factor has been related to the more familiar vibration transmission coefficient. A full scale measurement program was conducted on the Sikorsky S-76. Data are presented showing the SEA parameters and comparisons with the analytic model. These data include vibration transmission along the beam/panel structure and transmission through intermediate acoustic spaces connected by both resonant and nonresonant conditions. In addition, data are presented concerning the effects of coherent and noncoherent source excitations at distant response points. [Work supported by NASA Langley.]
Point-force random excitation of a thin plate bent into an L-shape
Authors: Dimitriadis Emilios K. , Pierce Allan D.
Published: January 2006

The authors' model for random point-force excitation [J. Acoust. Soc. Am. Suppl. 1 73, S71 (1983)] is refined and simplified. The modal cross-correlation terms introduced by point forces are accomodated by using the method of images with only the nearest edge taken into account to approximate the power input to a finite plate. Arguments analogous to those used in room acoustics [see for example, Acoustics (McGraw-Hill, 1981), p. 295] allow the average cross-correlation term to be expressed in terms of the power input ratio, finite plate to infinite plate. Statistical energy analysis (SEA) parameters are computed for the L-configuration plate for the cases when all edges are simply supported and when all edges are free, with Warburton's Rayleigh-Ritz modes used in the latter case. The analysis confirms that the SEA parameters are very nearly independent of force position. Calculations compare favorably with experiments conducted on a freely suspended L-configuration plate with different source locations and two different damping configurations. [Work supported by Whirlpool Corporation.]
An acoustic direction finder
Authors: Coker C. H. , Fischell D. R.
Published: January 2006

A simple procedure is described for finding the direction to the source of an acoustic signal—primarily the human voice. The method, in effect, constructs the cross correlation of severely distorted signals from two microphones. The distortion, a combination of time-domain processing and digital logic, is such to give strong preference to direct-path early arrivals over reflections that follow, and to produce streams of pulses whose cross correlations can be computed very simply. The method has been implemented as a combination of custom hardware, and a simple microprocessor. The implementation operates in real time (with ∼0.2-s delay) and produces outputs to indicate either the azimuth angle, or inter-microphone delay. Possible uses include self-aiming directional microphones, automatic aiming or switching of video cameras, and remote identification of talkers in nonvideo teleconferencing.
Asymmetrical frequency discrimination in primates
Authors: Sinnott Joan M. , Petersen Michael R.
Published: January 2006

Auditory frequency difference limens at 0.5, 1.0, 2.0, and 4.0 kHz were measured monaurally in humans and in four Old World monkey species: Japanese and rhesus macaques (Macaca fuscata, Macaca mulatta) and vervet and deBrazza monkeys (Cercopithecus aethiops, Cercopithecus neglectus). A go, no-go, repetitive-standard discrimination procedure was employed with positive reinforcement. At certain frequencies and sensation levels, humans and macaques exhibit superior discrimination for frequency increments, while under these same conditions the Cercopithecus species exhibit superior discrimination for frequency decrements. Possible coding mechanisms to account for these results will be discussed, including monitoring rate increases versus rate decreases in asymmetrically tuned auditory nerve fibers. In one species, the Japanese macaque, there is some evidence that superior discrimination of frequency increments may relate to the design features of the animal's vocal communication system. [Supported by NSF, Sloan Foundation, and Deafness Research Foundation.]