bluuebunny/crossref_metadata_2025 huggingface dataset:
Recognition strategies in a continuous speech understanding system
Authors: Mariani Joseph J.
Published: October 2005

ESOPE0, the first version of our speech recognition system, uses a top-down strategy from the pragmatic level to the phonetic one, and operates from left to right with a best-few method and no-backtracking. Dynamic comparison among the four best phoneme-candidates is carried out. ESOPE1 uses the same basic strategy in a systematic way: A best-few algorithm leads to a beam-search procedure. ESOPE1-1 employs a top-down treatment down to the acoustic level with a diphone dictionary. It uses a dynamic comparison method at the acoustic level. In our automatic dictation project, using a natural language syntax and a 170 000-form vocabulary, a bottom-up, best-few attitude has been taken to translate into words an error-free continuous phoneme string. We therefore feel that severely limited language and poor phoneme recognition involve a top-down strategy, whereas a bottom-up strategy is preferable in the opposite situation. This, and the recent results in psycholinguistics, lead us, in our present elaboration of ESOPE2, to the use of both a top-down, and a bottom-up strategy (Prediction-Verification-Induction). Predictions are made at each level, but the recognized phonemes may introduce unpredicted words, to allow limited learning abilities.
The use of syntax, semantics, and pragmatics in the KEAL speech understanding system
Authors: Gillet Dominique , Nouhen-Bellec Andrée , Quinton Patrice , Siroux Jacques
Published: October 2005

KEAL is a speech understanding system currently under development at the French Telecommunication Research Center (CNET) in Lannion; it aims to study oral man-machine communication. KEAL is typically designed for automatic inquiry. Such a task requires dialog procedures in order to provide naturalness in the inquiry process. On the other hand, the use of dialog allows to achieve good comprehension despite the presently limited performances of KEAL at the phonetic level. This paper describes how syntactic, semantic and pragmatic knowledges are used in KEAL. Sentence recognition is performed by a bottom-up left-to-right parser which provides also the parse-tree of the sentence. This parse-tree is then interpreted in order to extract the semantic information which is relevant to the dialog. The semantic information is used by the dialog controller for instantiating a model-graph, which represents the state of the dialog at any moment. The dialog controller sends a response to a text-to-speech synthesizer, and indicates how to analyze the user's reply. An example taken from automatic directory assistance service is described; results concerning preliminary experiments are discussed.
On the use of a speech recognition system for the detection of mispronunciations
Authors: Le Bras Jane
Published: October 2005

The investigation described in this article formed part of a thesis whose subject relates to the use of a speech recognition system in the field of automatic detection of mispronunciations. The experiment presented here is a first attempt at automatic evaluation of errors of pronunciation made by English subjects learning French. After briefly describing previous work using computers in the teaching of pronunciation, problems connected with the choice of a speech-recognition system will be discussed. Then, we will describe the method chosen and the constraints specific to the using of an analytic speech-recognition system. An algorithm for detecting the presence or the absence of aspiration in stop consonants will be presented. In the last part of this work, we will give the results of experiments carried out with five English subjects and seven French subjects using a program employing this algorithm. This program is simply intended to facilitate the testing of algorithms and to allow research with several different speakers. It has not been written in real time, but in batch (IRIS 80). Its principal interest comes from the fact it uses the output of the phonetic analyser of keal, the speech recognition system used in CNET.
Computer controlled measurements with a spherical resonator
Authors: Barrett Gültepe M. A. , Gültepe M. E. , Yeager E. B.
Published: October 2005

A microcomputer controlled spherical resonator apparatus has been developed for acoustic studies of liquids at 5–500 kHz. A programmable frequency synthesizer producing frequencies in steps down to 1 μHz is used as the source. The received signal is fed into a programmable tuned amplifier, and demodulated. A/D conversion is made with a high-speed sample-and-hold digital voltmeter via a multiplexing scanner. The system allows for very detailed mode mapping of the sphere by accumulating amplitude data both while the sphere is being driven and after a specified decay time interval, along with recording of temperature. The experimental resonance frequencies have been compared with those predicted theoretically and decay rates of a large number of resonances measured. The amplitude decay can be followed through 70 dB with reproducibility within ±0.2 dB s−1 under favorable conditions. The system is also used for cylindrical resonators. [Research supported by ONR—Physics Program.]
Sensitivity of short-range SRBR transmissions to sound-speed profile selection
Authors: Bilazarian P. , Siegmann W. L. , Jacobson M. J.
Published: October 2005

The sensitivity of acoustic transmissions to the selection of a sound-speed profile model is analyzed when ray theory is valid. Profiles under consideration are depth dependent only and model the deep ocean sound channel. In this study, sound propagation is examined between fixed source and receiver located on the horizontal ocean boundaries and separated by less than 50 km. Given one profile, a procedure is developed for selecting a second simpler profile, for which all important ray-theoretic quantities are either identical or negligibly different. These include types and numbers of rays and per-ray phases and amplitudes. The selection criteria of the procedure are convenient to apply and are physically interpreted. Numerical illustrations are presented comparing profiles of bilinear and hyperbolic-linear types. Accurate formulas are derived which demonstrate the closeness of both per-ray and total-field phases and amplitudes for the two profiles. Thus, when the procedure is applied, propagation results will not be sensitive to which profile is used. [Work supported by ONR.]
Convergence zone macrostructure in a positive gradient environment
Authors: Walkinshaw H. M.
Published: October 2005

This paper completes a series of three studies on convergence zone structure in the deep ocean. Special properties of the positive (pressure) gradient sound speed profile, gp = 0.018 s, are described in the context of new definitions for a convergence zone and for depth excess proposed in two preceding papers: J. Acoust. Soc. Am. Suppl. 1 65, F5 (1979); and 66, JJ21 (1979). For example, for the profile considered here, reciprocal (conjugate) depth does not exist. All zones are simple, RSR convergence zones and multiply branched caustics do not form. In a sound-channel environment, a deep source couples efficiently into the channel and propagates to long range with substantially lower transmission loss than a shallow source. For a deepwater, positive gradient environment, the reverse is true. The deeper the source, the greater the transmission loss. The following calculations support the study: Ray diagnostic codes are used to construct source/ receiver ray angle θ and arrival order geometries between shallow and deep sources (18 and 152 m), and five receiver depths (18, 152, 305,610, and 1219 m). Corresponding changes in propagation loss with range are described by normal mode and parabolic approximation codes. All calculations are for 50 Hz over a high-velocity, sedimented bottom at 2100 fms (3840 m). CZ internal composition is also analyzed in terms of contributions of discrete groups of modes to overall propagation loss.
A comparison of measured arrival structures with several theoretical models
Authors: Freese Herbert A.
Published: October 2005

Multipath arrival structures were measured using short pulse transmissions in various regions of the North Atlantic Ocean in the presence of a surface duct. These data were compared with the predictions obtained using the parabolic equation method, the multipath expansion method and classical ray techniques. Particular attention is drawn to the importance of energy which leaks through the duct and into bottom bounce or refracted modes and also the reverse leakage situation. It is pointed out that in some instances these “leaky” modes contain most of the received energy and that classical ray techniques must be avoided in this case. [Work supported by NAVSEA.]
Comparisons of the ducted precursor model with data and fullwave model results
Authors: Keenan R. E. , Spofford C. W.
Published: October 2005

The ducted precursor model described in the preceding paper is compared with numerical results generated using a broadband version of the parabolic-equation model, and with data collected at sea. While ducted precursors have usually been observed in convergence zone environments, the data were collected in a bottom-limited environment and exhibit an arrival structure quite different from the earlier data. When not recognized as precursors, these arrivals have in the past been mistaken for purely bottom-bounce paths of different orders. There are two unique ways in which the data clearly differ from bottom-bounce arrivals: (1) the ducted precursors arrive equispaced in time rather than with an increasing separation; and (2) later arrivals are often of greater amplitude than the preceeding arrivals. These aspects of the data are shown to be well-represented by the model. [Work supported by NUSC/NLL and NORDA Code 500.]
The good acoustics of the Semper Opera in Dresden
Authors: Reichardt W.
Published: October 2005

The Semper Opera in Dresden had, before burning down in 1945, a distinguished reputation, among other things for its acoustics. Before starting its construction recently, the essential characteristics were defined by means of an acoustical model. These were (a) the inner surfaces of the stage opening; (b) wide, shell-shaped structures underneath the balcony fronts; (c) reduced depth of boxes, to obtain reflections from their rear walls; (d) a large, reverberant space above the fourth balcony. The reconstruction of the Opera house reproduces these characteristics almost accurately.
A method to improve the listening conditions in audience areas shielded from the main volume of the hall
Authors: Figwer J. Jacek
Published: October 2005

The listening conditions for audience located under deep balconies in halls can be improved through application of electronic systems providing a uniform distribution of the reverberant sound field. Such systems include microphones for pickup of reverberant sound in the main volume of the hall, electronic delays, and loudspeakers distributed above the audience area. Subjective listening tests during adjustments of actual installations indicate that the level of reverberant sound radiated from the loudspeakers must be approximately equal to the level of the reverberant sound from natural sources, at the same listening positions, for the systems to be acceptable and effective. Similar smaller scale systems are helpful to improve the listening conditions in sound control rooms with inadequate connections to the main volume of the hall.
Experiments on the variation of the acoustical properties of auditoria
Authors: Bradley J. S.
Published: October 2005

The acoustical design of an auditorium requires the prediction of its acoustical properties from architectural information. Although this may be possible for reverberation times, it is not possible to precisely predict values of quantities such as reverberant to early sound ratios and interaural cross correlations. The problem has been studied by measurements in a variety of real auditoria, and by calculations using computer models. A measurement scheme was developed that used only simple conventional equipment in the field and computer analysis in the laboratory. Measurements were thus readily obtained in a number of halls ranging in volume from 900 to 31 000 m3. Both within-hall, and between-hall differences were related to architectural details where possible. Computer models, using image source techniques, allowed the systematic variation of architectural parameters of simple room shapes. An initial understanding of the influence of gross architectural features on acoustical properties was obtained.
Simulation modeling of reverberation using a modified ray method
Authors: Wayman James L. , Vanyo James P.
Published: October 2005

The computer simulation technique for modeling acoustic fields, as developed over the last twenty years by numerous researchers (Allred and Newhouse, Schroeder, Haviland and Thanedar), has been modified to consider phase relationships between the numerous direct and reflected ray tubes. After verification using free-field and steady-state conditions, the model was applied to a theoretical reverberant enclosure. Simulated decay curves agreed well with both Eyring's equation for average decay rate and Maa's theoretical calculation [J. Acoust. Soc. Am. 18, 134–139 (1946)] for rms percentage fluctuation during decay.
The design and development of Godot: A system for computer-aided room acoustics modeling and simulation
Authors: Walsh John P. , Dadoun Norman
Published: October 2005

Godot is a software system for computer-aided room acoustics design and simulation. It is able to generate an audible simulation of arbitrarily directional sound sources in arbitrarily shaped rooms, incorporating diffraction effects. This paper reviews the design of the system. A polyhedral representation is used to model the room under consideration. Sound beams, defined by their axis directions and directivity factors, are traced as they traverse the room, and their “hits” upon a receiver position are noted. A perspective projection from the image of the source onto a “window” perpendicular to the beam axis is used to determine which edges of the room fall within a beam and hence can generate diffracted waves. The power transfer function of each path from source to receiver is used to derive the autocorrelation coefficients associated with the path, and linear least-squares prediction is used to determine the coefficients of an autoregressive digital filter which will match the frequency response of the path. When driven by arbitrary digitized program material, the parallel channels of the system, representing the paths to the receiver, will give an audible simulation of the conditions being considered.
Spectral similarities of tones from “especially useful” musical instruments
Authors: Benade A. H.
Published: October 2005

In a room the ear can track pitch, tone color and dynamic nuances of musical instrument tones with quickness, stability, and precision, even in the presence of many similar tones or of noise (otherwise music would be impossible). It has been recognized that the ear is most effective with tones consisting of a set of harmonic components; with fairly strong lower partials and upper partials that rapidly fall away. This spectral rolloff apparently reduces intercomponent masking in the upper critical bands, a phenomenon that exists above the 6th harmonic. A study has been made of room-average spectra for tones from instruments recognized as being especially useful in ensemble playing (oboes, violins, trumpets, pianos, saxophones, and bassoons). The spectra are remarkably similar: After a few srong lower harmonics each instrument shows an average rolloff of very nearly 18 dB/octave beginning at the 5th or 6th harmonic. There is no reason in physics why these diverse instruments share such spectral features. Instruments that “carry” less well (e.g., flute, cornet, or “vibes”) roll off earlier and/or more steeply. Tones with slower rolloff (e.g., the lowest note of an oboe sans lower joint) also show weakened trackability and pitch. [Work supported by NSF.]
Bassoon spectrum transformation function
Authors: Benade A. H. , Carman J. C. , Barrett P. H.
Published: October 2005

Individual notes and also musical passages have been recorded with one microphone moved about the room, and the other probing the bassoon's reed cavity signal. Room-averaged (“external”) spectra p‖(ext) and mouthpiece (“internal”) spectra p‖(lnt) are measured for the harmonic components of the mf level notes C2, F2, C3, F3. The precision is ±2 dB from all causes (player variability and room fluctuations). The smoothed transformation function 〈T(f)〉 = 〈pext/plnt〉 is calculated as a continuous function of frequency. This has a broad peak (30-dB high, 500-Hz wide at 15 dB down) centered at 600 Hz. Above this it fluctuates smoothly before rolling off steeply above 2000 Hz (20 dB down by 2500 Hz). The T‖'s of individual components differ from the corresponding values of 〈T(f)〉 by about ±6 dB. Loudspeaker playback of music using the internal signal via an electronic equalizer set to 〈T(f)〉 sounds like a bassoon but lacks “life.” Boosting or cutting several randomly chosen ⅓ octave bands by 4–8 dB improves the realism considerably, in the judgement of bassoonists and others. [Work supported by NSF.]
Cognitive correlates of musical similarity in chords for musicians and nonmusicians
Authors: Beal A. Lynne
Published: October 2005

The perceptual and cognitive structures for music recognition possessed by trained musicians and musically naive listeners were examined through a similarity judgement task. Listeners rated chord pairs for perceived musical similarity. Chord pairs were identical or different in musical structure. Timbre, an acoustical property not related to musical structure, was manipulated independently by playing chords on harpsichord and guitar. Chord pairs involved the same or different instruments. Musicians' similarity ratings were unaffected by timbre differences. Attributes related to the musical structure of the chords (e.g., pitch equivalences and component intervals) determined their ratings. In contrast, nonmusicians did not separate musical similarity from acoustical similarity and used different criteria for rating chord pairs played on same than on different instruments. These results support earlier findings (Beal and Allard, Canadian Psychological Association Convention, June, 1980) that musicians develop cognitive structures for analyzing musical stimuli through formal study of music theory. Without this expertise, nonmusicians must process the sounds on a more cursory acoustical level.
Tweeter loudspeaker sound diffusion by oblique reflection
Authors: Carlisle Richard W.
Published: October 2005

The radiation of sound over a wide angular beam has long been one goal of loudspeaker design. This problem becomes exacerbated as the frequency is increased, since the ratio of effective source diameter to wavelength increases. This problem has been approached by the use of 45° reflectors in the sound beam, by sectoral divisions in a horn, by reflective sections in folded horns, and the like. In the present instance, a pair of reflective surfaces is arrayed in the sound beam at an oblique angle to each other and at an acute angle to the axis of the beam. The sound beam is thus deflector at an acute angle to its original axis from each of the reflective surfaces. The beam is thus divided and diffused. Results indicate that the width of the beam can be increased by a factor of more than two. A model will be shown.
Control of transformer noise at Ontario Hydro
Authors: Edwards A. T.
Published: October 2005

The general philosophy is reviewed for controlling transformer hum in the community to acceptable levels. The various considerations in the design of acoustic treatments e.g., vibration isolation, walls combining high transmission loss and absorption, short ventilation ducts with high insertion losses, and the isolation of oil lines are discussed. Treatments for large units up to 750 mVA 500 kV working, having insertion losses up to 30 dBA are described.
Vibratory and acoustic response of systems subject to impact loads
Authors: Akay Adnan
Published: October 2005

A general theoretical analysis of the vibratory energy acquired by systems following impact is developed. Acoustic and structural losses are studied to predict sound radiation from impacted structures. Acoustic energy radiated from impacted beams is considered to predict noise characteristics of high speed machine elements. The analytical results relate the time rate of change of the contact force history to the vibrational energy of the impacted structure and to the radiated acoustic energy. External energy losses are related to the radiation efficiencies in describing the acoustic energy. [Work supported by NSF and WSU.]
Nonlinear image processing for precise localization of noise sources
Authors: Sato Takuso , Sasaki Kimio , Nishio Naoki , Ishida Masaya
Published: October 2005

The localization or imaging of abnormal parts of machine systems from the noises detected by microphone array is desired in many fields of nondestructive diagnoses [T. Sato, K. Uemura, and K. Sasaki, J. Acoust. Soc. Am. 67, 1802–1808 (1980)]. The random nature of the machine noises and the short length of the available data restricted from the dynamic detection of the machine's state, however, results in fairly significant specklelike noises in the reconstructed images. In this paper, a few nonlinear processings of the obtained images are considered aiming at precise localization of objects even from these noisy images. First, a binary mask is produced from the observed image itself by comparing the intensity at each part with a certain level, then the correlation image between the mask and the original image is obtained [R. H. T. Bates, M. O. Milner, G. I. Lund, and A.D. Seagar, Opt. Commun. 26, 22–25 (1978)]. The resulting images give a fine distinction of even neighboring objects suppressing the speckle noises. The formulation of the method, numerical results, and some experimental results are given.
Acoustical source—Ranking of transformers using vibration analysis
Authors: Ellingson Edward F.
Published: October 2005

In order to properly approach the problem of reducing the noise level of a transformer, the transformer as a noise source, must be thoroughly understood. A technique is presented for source-ranking transformers using vibration data. The technique was developed and demonstrated during a contract between Allis-Chalmers and the Electric Power Research Institute (EPRI). The technique uses a summed acceleration signal from accelerometers placed on small areas of the transformer and is processed via a desktop computer controlled fast Fourier transform analyzer. When combined with the radiation efficiency of that area, calculated from physical properties, a sound-power level for that area is calculated. The sound-power level for any of these small areas can then be added together, in any combination desired. Trade-offs between cost and reduction required can then be intelligently made.
Spatial statistics of transformer core noise
Authors: Vér István L.
Published: October 2005

The tonal nature of the core noise, the difference in pathlength and the different phasing of the sound radiated from the various tank surfaces result in complex spatial interference patterns which characterize the farfield which can be described only in statistical terms. This paper deals with the consequences of the statistical nature of sound field in regard to: (1) Developing meaningful techniques to measure farfield transformer core noise, (2) predicting the expected value and spatial statistics of the farfield noise on the basis of near/field measurements, and (3) defining meaningful performance evaluation measures to determine the effectiveness of such retrofit noise control measures as barriers and enclosures.
Invariant acoustic correlates of phonetic features and their potential role in speech perception
Authors: Stevens Kenneth N.
Published: October 2005

A version of the theory of acoustic invariance states that are invariant acoustic correlates of each phonetic feature, independent of the context in which the feature appears. Examples from a number of features are given to show that, if the appropriate acoustic attribute is defined for each feature, there is indeed a high degree of acoustic invariance for the feature. The implications of such a theory for models of language acquisition and speech perception are discussed. It is suggested that a realistic model of speech perception must have the capability to identify lexical items both through the extraction of a pattern of invariant acoustic attributes for individual features in the matrix that defines the item and, in situations where these attributes are obscured, through the use of secondary, context-dependent cues. [Supported in part by a grant from NINCDS.]
Long-range underwater acoustic communciations—A review
Authors: All Hassan B.
Published: October 2005

Long-range underwater acoustic communication systems are constrained by the properties of the medium. In particular, the non-uniform (frequency-dependent) attenuation of the signal amplitude, coupled with multipath and Doppler-induced time and frequency spreading, limit long range communications to low frequencies (less than 500 Hz), relatively low data rates (of the order of bits per second), fairly narrow frequency bands. Methods to overcome, or mitigate, the medium-induced signal degradations are based on attempts to optimize the waveform/receiver structures to the characteristics of the propagation channels. Implicit diversity, provided by convolutional coding and interleaving, frequency-hopping spread spectrum techniques, and Adaptive RAKE-like receivers provide considerable signal protection against multipath fading and interference.
Vibration characteristics of a floor sample
Authors: Rainer J. H. , Pernica G.
Published: October 2005

Vibration tests consisting of heel impacts by a person, shaker impacts, and random shaker excitations were carried out on an 8.5 × 8.3 m steel-joist-concrete floor. Structural variations to the floor sample included edge conditions restrained and free; two slab thicknesses; changes in joist stiffness; and inclusion of transverse bracing. The lower four to six natural frequencies and associated mode shapes and damping values are determined. Damping for the sample is significantly increased by the presence of one or more persons on the floor. Viscous damping coefficients for people are evaluated, and comparisons presented between measured and calculated damping ratios, modelling people by simple dashpots. Predicted and measured acceleration responses due to heel impact, shaker impact, and shaker white-noise excitation are also compared.
An improved transducer mount for ground vibration measurements
Authors: Barman M. , Coulter J. E.
Published: October 2005

A simple, convenient, and inexpensive transducer mounting system for soil vibration measurement is suggested. Analysis indicates excellent transducer coupling to soil in all axes of motion. Testing to date indicates an improvement of about two octaves in vertical frequency response over common transducer mounting techniques. At the same time the device avoids some of the shortcomings of other mounting methods.
The effects of ultrasound on the electrophysiology of cardiac muscle
Authors: Mortimer Alan J. , Roy O. Z. , Bresden Boris J. , Forester George V.
Published: October 2005

Using a system developed at the National Research Council of Canada the effect of ultrasound on cardiac muscle electrophysiology and on contractile properties of the myocardium has been studied. Preliminary results have been obtained from seven rat papillary muscles. Ultrasound led to a decrease in resting tension of the papillary muscles which paralleled a cellular hyperpolarization of the resting potential of 3 to 5 mV. In addition ultrasound led to an increase in the overshoot of the action potential of 26 mV. Observed changes in overshoot may be explained due to temperature changes, however changes in resting potential cannot.
Loudspeaker cone behavior: A brief tutorial
Authors: Kates James M.
Published: October 2005

Loudspeakers are often described by a lumped-parameter model; the major components of this model are the cone mass, the suspension stiffness, and the motor and mechanical damping. Such a simplified model is adequate at low frequencies, but at high frequencies the vibrational behavior of the cone itself comes into play as well. This causes various normal modes to occur in the cone or dome motion, often termed cone “breakup.” In this paper, we will review the mechanisms and effects of loud-speaker cone and dome behavior, and we will discuss some of the techniques used to overcome this problem in loudspeaker design.
New materials for loudspeaker diaphragms and cones—An overview
Authors: Yamamoto Takeo , Tsukagoshi Tsunehiro
Published: October 2005

Desirable properties of materials for loudspeaker diaphragms are high specific Young's modulus (E/ρ), large flexural rigidity and large internal loss (tanδ). In order to obtain large flexural rigidity, honeycomb structured diaphragms or porous metal diaphragms have been used. For the purpose of realizing ideal loudspeaker diaphragms which can be easily produced, however, the material itself should have high values of E/ρ and tanδ This paper is a review on recent developments of materials for loudspeaker diaphragms in Japan, such as beryllium, boronized titanium, carbon fiber reinforced olefin, and polymer-graphite composite diaphragms.
Sound exposure level of a sonic boom from its discrete Fourier transform
Authors: Young Robert W.
Published: October 2005

In a usual digital spectrum analyzer, a transient such as a sonic boom is captured by taking 1024 samples during a time window T. Under the name of discrete Fourier transform, the coefficients of a Fourier series of period T are computed, the square of the absolute value of each constituting a time-window mean-square sound pressure associated with a “bin” whose frequency width is b = 1/T Hz. The output is often incorrectly described as a power spectrum, whether the variable is sound pressure, voltage, or whatever. The whole analysis band is B = 400/T = 400/b Hz. Sometimes “total power” is calculated as the sum of the squared bin sound pressures. A time-mean square is an ambiguous measure of a transient, because the averaging time is an arbitrary artifact of the measuring instrument. The solution is to multiply the mean-square sound pressure by the time window, to obtain the time integral of squared sound pressure; and to call it sound exposure or voltage exposure, etc. Examples as levels customary in acoustics are, with T = 400 ms, B = 1 kHz, b = 25 Hz the 400-ms average. 2.5-Hz bin sound-pressure level at 125 Hz was 103 dB; the 1-kHz band sound-exposure level was 134 dB.
The effects of flow and geometry on the acoustic characteristics of bends
Authors: Cabelli Andre , Shepherd Ian C.
Published: October 2005

A two-dimensional finite element technique was used to investigate the acoustic characteristics of 90° bends for plane wave and for cross-mode propagation. An experimental procedure was also developed which permitted measurement of these characteristics in terms of rigid walled duct modes when several higher order modes were admitted simultaneously. Numerical and experimental results displayed excellent agreement over a range of frequencies extending to the cut-on value of the third cross mode. Characteristics were studied in terms of modal amplitudes of the reflected and transmitted waves and in terms of the corresponding energy coefficients. The energy transfer was also examined for the case when an evanescent mode was incident on a mitred bend. The influence of changes in geometry was established over a range of inner and outer radii and the effects of locating a turning vane in the bend were also examined. The energy reflected was related, where possible, to published aerodynamic characteristics. The effects of a mean compressible flow on the acoustic characteristics was examined for one geometry.
Measured performance and predicted transmission loss of plug mufflers
Authors: Noreen Robert A. , Hoops Richard H. , Anderson Craig A.
Published: October 2005

Predicted transmission loss characteristics of plug mufflers are compared to measurements of both transmission loss and insertion loss. A plug muffler is a cross flow expansion and a cross flow contraction perforated tube element connected in series. The predictions were based on transfer matrix calculations using a decoupling approach which gives closed form expressions for the matrix terms [K. Jayaraman and K. Yam, J. Acoust. Soc. Am. 69, 390–396 (1981)]. Transmission loss measurements with and without flow were made on an impedance tube with a horn-driver as a noise source. Insertion loss measurements were made on the impedance tube, a cold flow facility, and on an engine. Results (zero) are shown for Mach numbers from 0–0.18 at ambient and elevated temperatures. The agreement between the measured and predicted results is excellent.
The spectral shape of the dominance region for ripple noise pitch
Authors: Yost William A.
Published: October 2005

The repetition pitch and pitch strength of ripple noise depends on information in a spectral region located three to five times the reported pitch. A quantitative measure of the shape of this spectral dominance region for repetition pitch was obtained using a pitch strength, discrimination procedure. Listeners discriminated between two pitches of ripple noise. The pitch strength of the two stimuli was decreased until the listeners were at threshold in their ability to make the pitch discrimination. The pitch strength required for threshold was determined as a function of low- or high-pass filtering the ripple noise using filters with a variety of cutoff frequencies. This function described the spectral dominance region for a variety of ripple noise conditions. The dominance region had a bandpass characteristic with a center frequency at four times the reported pitch, and the low-frequency roll off in pitch strength was shallower than the high-frequency roll off. The shape of the dominance region was proportional to the repetition pitch of the ripple noise. Thus, pitch strength was strongest when the spectrum contained energy at four times the reported pitch, weaker when only high-frequency energy was present and weakest when only low-frequency energy was present. These results will be discussed in terms of the recent models of pitch perception, especially the Peripheral Weighting Model. [Work supported by NSF.]
Effects of phase on the perceived pitch of the higher component of a two-tone harmonic complex
Authors: Bundy Robert S. , Colombo John
Published: October 2005

Subjects were presented with a 1000-Hz tone of 1-s duration during which two 2000-Hz tones were sequentially interspersed. Each of these higher tones varied in their phase relation with the lower tone from 0° (cosine coincident) to 315° in 45° increments. All possible phase pairs were presented to the subjects. For each trial the subjects were asked to judge whether the second interspersed tone was higher or lower in pitch than the first. Subjects were able to produce orderly data indicating that the phase relationship of the higher tone reliably affected the perceived pitch. The phase relationship that produced the highest pitch varied both between trials in particular subjects and across subjects. Across all trials and subjects no particular phase relationship was significantly more likely to be judged as having the highest pitch.
A model of the perceptual processing of rhythmic auditory pattern temporal order
Authors: Wood Robert W. , Chial Michael R.
Published: October 2005

Studies of auditory rhythm are complicated by difficulties in describing the features of tonal sequences that influence perception. One such feature appears to be redundancy. A model is proposed which describes (1) an analytical procedure to quantify the redundancy of temporal relations among elements of temporal patterns, and (2) an explanation of how redundancy may affect the perceptual encoding, decoding, and recoding of rhythmic patterns. The analysis procedure yields indices of total possible redundancy for classes of patterns, indices of the redundancy of particular patterns, and ratio indices of relative redundancy which are largely independent of pattern class and which allow direct comparison among different patterns. Auditory perception is approached from the standpoint of redundancy reduction. In general, redundancy reduction allows for optimum efficiency in stimulus information transmission due to increased informational capacity of the perceptual system. It is proposed that rhythmic patterns displaying higher relative redundancy are processed perceptually with greater facility than less regular patterns for which redundancy reduction is minimized.
Barn owls can use ongoing-time disparities to determine the azimuth of a sound source
Authors: Moiseff A. , Konishi M.
Published: October 2005

A class of auditory neurons in the owl's nucleus mesencephalicus lateralis pars dorsalis (MLD), a midbrain auditory nucleus, have small spatial receptive fields. We sought our relationships between the spatial receptive fields of these neurons (measured under free-field conditions) and sensitivities to binaural disparities (presented dichotically). To switch easily between free-field and dichotic stimulation we constructed threaded, miniature earphones, which could be inserted into threaded rings implanted into the owl's ear canals. Neurons with spatial receptive fields were found to be sensitive to a narrow range of ongoing-time disparities in the biologically significant range of tens of microseconds. There was a strong correlation between the azimuth of the spatial receptive fields and the magnitude of the ongoing-time disparities. Freely behaving owls were presented with similar ongoing-time disparities through identical earphones. They reacted to the stimuli in a manner consistent with the conclusion that ongoing-time disparities were used to ascertain the azimuth of a sound source.
Pitch perception in musicians is altered by short-term experimental manipulation
Authors: Bundy Robert S. , Colombo John , Warnick Patty
Published: October 2005

Subjects were tested for pitch matching before and after a one-hour exposure to seven component tones in which the first three harmonics were either sharp or flat by 3%. In the test for pitch matching the subjects were asked to adjust a tone consisting of harmonics 1–3 so that it matched the pitch of a tone consisting of harmonics 4–7. Comparison of the pre- and post-exposure matching performance indicated that subjects were exposed to the overtones that were tuned sharp matched higher while the subjects exposed to the overtones that were tuned flat matched significantly lower on the post-test. This data suggests that normal pitch perception may be an acquired skill which is not necessarily dependent on the periodicities of the complex waveform or the integer relationship of the harmonics.
Normal threshold of hearing by bone conduction in an extended frequency range
Authors: Richter Utz , Brinkmann Klaus
Published: October 2005

Threshold measurements with a Präcitronic KH 70 bone vibrator (contact area 1.75 cm2, force of application 5.4 N) were carried out on a group of 25 otologically normal young subjects (50 ears) in the frequency range between 125 and 8000 Hz. The vibrator was applied to the human mastoid as well as to the forehead. The nontested ear was masked according to the rules given in ANSI S 3.13-1972. Equivalent threshold force levels were determined on an artificial mastoid type B&K 4930 of recent production. Between 125 and 2000 Hz the equivalent threshold force levels for mastoid application decrease steadily by about 13.5 dB/octave, they then increase again by about 10 dB and remain constant between 4000 and 8000 Hz. The equivalent threshold force levels for forehead application were found to be higher by about 11 dB in the whole frequency range. This is in contrast to results obtained by other authors during the sixties, who reported decreasing mastoid-forehead differences up to 4000 Hz determined with a Radioear B 70-A vibrator. This vibrator does not, however, provide the flat contact area of 1.75 cm2 required nowadays.
Detection of range motion in bat sonar
Authors: Simmons James A.
Published: October 2005

Echo-locating bats (Eptesicus fuscus) were trained to detect the presence of sonar echoes in a background of broadband random noise interference (20–100 kHz). The bats can detect echoes that are as much as 5 dB less intense if the echoes are shifting in arrival time instead of stationary. Measuring detection thresholds as a function of the amount of echo time-shift generates masking functions that correspond to the period structure of the bat's sonar transmissions and echoes. Shifts of 10 or 20 μs yield lower detection thresholds than shifts of 0 or 30 μs. The results indicate that the sonar receiver in broadband FM echo location may display motion in range as an independent target feature, and they confirm observations from range-jitter discrimination experiments that bats using broadband echo location perceive the phase of sonar echoes relative to transmissions. [Work supported by NSF.]
Neural and behavioral studies of auditory habituation in bullfrogs
Authors: Megela Andrea L. , Capranica Robert R.
Published: October 2005

The time course of habituation in response to pure tones was examined at three levels of the auditory nervous system in the bullfrog, Rana catesbeiana. Single-unit activity in the eighth nerve showed no evidence of habituation, even at stimulus repetition rates as rapid as one every second (stimulus duration 300 ms). Multi-unit and evoked potential activity from the torus semicircularis in the midbrain exhibited habituation decrements only at repetition rates of one every second or faster. Evoked potentials from the central nucleus of the dorsal thalamus showed rapid decrements in amplitude at repetition rates of one every ten seconds or faster. Maximal response amplitudes were maintained at repetition rates of one every twenty to thirty seconds. Latencies of the neural responses did not vary with repetition rate at any level of the auditory pathway. Adult male bullfrogs maintained in a seminaturalistic environment reliably produced mating calls in response to taped conspecific calls when playback repetition rate was one every twenty to thirty seconds. These results suggest that acoustical processing in the dorsal thalamus plays a significant role in mediating the evoked calling behavior of the bullfrog. [Supported by NIH fellowships to A.L.M. and NIH grant NS-09244 to R.R.C.]
A psychophysical estimate of the traveling-wave delays in the human cochlea
Authors: Schwartz J. L. , Escudier P.
Published: October 2005

Recent experiments using physiological means (such as recordings of action potentials in electrocochleography or studies of brainstem evoked responses to click stimuli) have allowed the accumulation of a first set of results on traveling-wave delays in the human cochlea, but very few psychophysical results are available. We propose here a systematic psychophysical study. Two methods have been employed, both using pairs of short (2 ms) high-frequency (F ⩾ 1000 Hz) stimuli, the frequency of the higher one (the reference) being fixed at 8000 Hz. Both methods are based on the determination of the advance that the lower frequency sound must have in order to make up for its longer travel time in the cochlea so that a subject will perceive the two sounds as simultaneous. Finally, we can propose a psychophysical estimate of traveling-wave delays in the basal part of the human inner ear. The results are in very good agreement with those obtained in physiology.
The perception of sound in association with auroras
Authors: Roederer Juan G.
Published: October 2005

The aurora is a large-scale, dynamic optical phenomenon occurring at 80–100-km altitude and high geomagnetic latitude, caused by energetic electrons precipitating into the upper atmosphere. No sound waves in the audiofrequency domain can propagate from auroral altitudes to the ground, and none have ever been physically detected. Yet there are numerous reports of auroral “sounds” perceived during auroral displays [see review by S. M. Silverman and T. F. Tuan in Adv. Geophys. 16 (1973)]. While some reports may have an obvious explanation related to environmental sounds, we propose that auditory sensations during auroral displays are the result of an illusory associative recall, or the consequence of a synaesthetic effect: when a large portion of the retina is stimulated with coherently moving patterns (as in the aurora), auditory associations may be triggered by optical-acoustic channel interaction, perhaps at the superior colliculus level. This interaction may be enhanced during particular behavioral states, such as usually prevailing during the observation of aurora (darkness, cold, silence, isolation).
Interactions of lexical status and stimulus dominance effects in dichotic listening
Authors: Paul-Brown Diane , Soli Sigfrid D.
Published: October 2005

Lexical status and stimulus dominance effects in dichotic listening were examined. Stimuli were naturally produced CVC words and nonwords equated for amplitude and duration. Dichotic pairs varying in lexical status and/or initial consonant voicing were recorded in blocks of trials. Groups of eight male and eight female listeners were instructed to identify both stimuli on each trial. Responses for the two groups were analyzed separately for pairs differing on each stimulus dimension. Words were recognized 20% more accurately than nonwords when competing stimuli shared lexical status, and approximately 18% more accurately in either ear when competing stimuli differ in lexical status. Stimuli with initial voiceless consonants were recognized about 5% more accurately than voiced stimuli when pairs shared voicing, and 11% more accurately in either ear on trials when voicing contrasted. The magnitude of lexical status and stimulus dominance effects was approximately 7% greater for females. The pattern of these effects differed for the left and right ears and for males and females. These interactions are discussed in terms of linguistic processing mechanisms and general attentional strategies.
Dichotic “challenges” to CV syllables: The effects of transition duration
Authors: Porter R. J. , Lyman R. , Miller C. J.
Published: October 2005

Listeners have difficulty identifying stop + vowel CV syllables presented to one ear when various other types of signals are simultaneously presented to the other ear. The signals which provide the greatest “challenge” to CV identification appear to be those which have “CV-like” acoustic properties (although the challenges themselves may not be perceived as CV's). Our work has been directed at isolating the extent to which different acoustic properties contribute to a signal's effectiveness as a challenge. Our objectives are to further clarify the dichotic interaction of complex signals and probe the nature of the processes which analyze the acoustic cues of speech. Our previous work suggests the presence of a transition together with a final steady-state are important attributes or effective challenges; the duration of the steady state also seems important. Steady-state frequency and transition onset-frequency, on the other hand, appear relatively less important. This study investigated transition duration variation over the range of 20–94 ms. Listeners identification of CV's (with 45-ms transitions) was found to be least disrupted by challenges with similar transition durations. This curious result, together with those of the earlier studies, suggest the disturbing effects of a challenge may be tied to the degree of disparity in the temporal distribution of acoustic energies in the two signals. This possibility will be further discussed in the context of our objectives. [Work supported by NIH-NINCDS NS1164706A1 and Kresge Foundation.]