bluuebunny/crossref_metadata_2025 huggingface dataset:
Components of lingual and labial articulation
Authors: Edwards Jan , Harris Katherine S. , Tuller Betty
Published: January 2006

One problem in speech production research is that it is difficult to relate muscle activity in individual articulators to the resulting movements. In large part, this is because movement of any given articulator is highly dependent on movement of connected structures. Both tongue blade and lip movements contain a component due to jaw movement in addition to a component that reflects muscle activity of the tongue or lip. Using data obtained by the x-ray microbeam system [Kiritani et al., J. Acoust. Soc. Am., (1975)], tongue blade and lip movements were separated into these two components. Dynamic characteristics of tongue blade articulation of /i/ and /a/ and labial articulation of /p/ were examined across multiple tokens of two to four syllable nonsense utterances. The timing of lip and tongue blade articulations relative to jaw raising for /p/ and jaw lowering for /i/ and /a/ is influenced by surrounding phonentic context and syllable stress. It was found that the lip and the tongue blade compensate for some but not all of the variations in inter-articulator timing and positioning. [Work supported by NINCDS.]
Digitizing effects on noise dynamic range and signal spectra in broadband arrays
Authors: Green M. D. , Young J. W. , Sullivan F. I. M.
Published: January 2006

Digitizing of the signals from elements in an acoustic array results in various types of errors in the beamforming process. Among these are quantization noise, clipping noise, spectral distortion, and false-target generation associated with clipping. This paper presents an analysis of these effects in the context of design requirements for a broadband receiving array. The array must operate over a specified dynamic range of ambient noise. A method is developed for ensuring that the sum of quantization noise and clipping noise will be kept at a specified level below ambient noise over the entire operating range. The optimum selection of quantization levels for this purpose is found. When a signal exceeds the saturation level of the A/D converter, spurious signals will be generated at new frequencies, which may alias back upon the band of interest. When two or more signals jointly exceed the saturation level, intermodulation effects among signals having different arrival angles may produce spurious signals which are coherent across the array. These signals appear in frequency-wavenumber space as false targets. An analysis and a computer simulation of this effect are presented. The requirements on the minimum acceptable saturation level of the A/D converter are determined.
Cross correlation of two hyperbolic-frequency-modulated (HFM) waveforms
Authors: Chou S. I.
Published: January 2006

In a multi-user radar/sonar environment, or in the case of a single user but with different pulses transmitted, it is important to know the cross correlation of the waveforms as a function of time delay. The cross-correlation property of two differently swept LFM (linear-frequency-modulated) waveforms is straightforward from the view point of using the stationary-phase approximation. There, as one waveform's frequency-time curve slides against the other's, the intersecting angle remains constant over an interaction interval in time delay corresponding to the common frequency interval of the two waveforms. The picture changes vastly when the two LFM waveforms are replaced by two HFM (hyperbolic-frequency-modulated) ones. In this paper, the cross correlation over such an interval is established. Using examples, the analytical approximation is shown to compare favorably with numerical simulation results. The difference between the analytic and numerical results is further characterized using the unity volume constraint for narrow-band cross-ambiguity function which is lesser known than the auto-ambiguity counterpart.
Coherent wavefront resolution
Authors: Murphy Donald A.
Published: January 2006

Resolving coherent arrivals is fundamentally different from resolving arrivals from independent sources because an interference field is created by the coherent arrivals. This results in a field which is not spatially stationary. At high signal-to-noise ratios considerably better resolution is possible than for the case of independent sources. An array which has a null of the interference field at its center will resolve the two arrivals for apertures considerably smaller than expected by using the Rayleigh criterion. The phase difference of the two arrivals is critical to resolution in this case. Resolution of this type was observed with signals which were received at a fixed array in the deep ocean from a single low-frequency source over long propagation paths through complicated sound speed structures, thus demonstrating the coherence of the multipaths involved. The coherence of the vertical multipaths follows from the discrete nature of the arrival structure. In this case each multipath is not appreciably spread in arrival angle.
Sensitivity of a passive horizontal-tracking algorithm to input-measurement errors
Authors: Bilazarian P. , Siegmann W. L. , Jacobson M. J.
Published: January 2006

We consider the passive determination of bearing, range, depth, frequency, and horizontal velocity for a narrow-band acoustic source which moves at constant depth. An algorithm for tracking a moving source is developed and employed, using a horizontal linear array which is towed at fixed depth. The source and receiver move with constant horizontal velocity in the upper portion of a deep ocean and are separated by relatively short range. Acoustic signals are presumed to arrive along two upper-ocean ray paths. The tracking is accomplished using a new combination of processed multipath information from the ray paths, frequency shifts from a cw emission, and directional meaurements from the linear array. The sensitivity of predictions for three-dimensional position, frequency, and horizontal velocity are demonstrated for small variations in environmental parameters and acoustic measurements. Results for a variety of source-receiver configurations are discussed. Variance estimates of position and motion predictions are obained in terms of input measurement variances. [Work supported by ONR.]
Coherent response to a point source irradiating a rough plane
Authors: Lucas R. J. , Twersky V.
Published: January 2006

We consider a point source of sound exciting a planar rough surface, and write the coherent response as the Sommerfeld-Weyl integral in terms of the reflection coefficient (R) for correlated distributions of protuberances (bosses) on rigid or free base planes. The coefficient R and the associated angle-dependent impedance are determined by the ensemble-averaged multiple scattering amplitude (F) for one fixed boss of revolution, and F is specified by its single scattered value and the statistical-mechanics radial distribution function (p) for impenetrable disks. Low-frequency forms in terms of simple integrals of p delineate multipolo-coupling and packing effects on propagation and attenuation (arising from incoherent scattering as well as absorption) as functions of the fraction (w) of base plane covered by bosses. Approximations of the response integral and corresponding plots are presented to exhibit the dependence on w and on other key parameters, and near-grazing aspects are emphasized.
Exploring allophonic and lexical constraints in a continuous speech recognition system
Authors: Chen Francine R. , Zue Victor W.
Published: January 2006

Recent research [Zue and Shipman, J. Acoust. Soc. Am. Suppl. 1 71, S7 (1982); Huttenlocher and Zue, this meeting] has shown that a broad phonetic representation of speech provides strong constraints for lexical access of isolated words. In addition, Church (Ph.D. thesis, MIT, 1983) has demonstrated the utility of detailed allophonic constraints for parsing a sentence from a phonetician's transcription. This gives reason to believe that the coupling of lexical and allophonic constraints can be a powerful tool in continuous speech recognition. The present study explores how broad phonetic constraints can be applied to a restricted continuous speech task. Using a broad phonetic representation derived from an ideal transcription, it was found that on the average, 70% of the word boundaries in the digit vocabulary can be identified. Extending this approach to speech data, we have implemented a classifier which derives a broad phonetic representation from the speech signal. Preliminary results indicate that allophonic and lexical constraints can be effective in reducing the number of string candidates, based upon the output from this classifier. [Work supported by the Office of Naval Research under contract N00014-82-K-0727 and by the System Development Foundation.]
The use of phonotactic constraints to determine word boundaries
Authors: Lamel Lori F.
Published: January 2006

Phonotactic constraints limit the permissible word internal consonant sequences in English. In some cases, knowledge of the phoneme sequence uniquely specifies the location of the word boundary, while in other cases, phonological rules based on allowable consonant sequences are not sufficient. For example, the word boundary can be uniquely placed in the sequence /…mgl…/, as in the word pair “some glass,” whereas the word boundary location is ambiguous in the phoneme sequence /…str…/ without further acoustic information. The /…str…/ may have a word boundary in one of three places as in “last rain”, “race trials”, and “may stretch.” Studies were conducted to determine the utility of phonotactic constraints to predict word boundaries. The data bases included the Merriam-Webster Pocket Dictionary, a phonemically balanced set of sentences, and samples of unrestricted text. Results indicate that: (1) word internal consonant sequences represent a very small subset of all permissible consonant sequences across word boundaries; and (2) acoustic-phonetic knowledge is needed when the word boundary is ambiguous. Results on the differences between word and syllable boundaries will also be presented. [Work supported by the Office of Naval Research under contract N00014-82-k-0727 and the System Development Foundation.]
Speaker normalizing transforms for automatic recognition
Authors: Sejnoha Vladimir , Mermelstein Paul
Published: January 2006

A method for the reduction of inter-speaker differences by spectral transformation will be described. Transforms were designed for the additive compensation and frequency scaling of speech spectra based on the log of the energies of 20 channels spaced on the mel-scale. The frequency scaling was implemented as either a linear scaling, or as a nonlinear frequency warp found by dynamic programming. The transforms were then combined so that the additive component principally addressed the differences in the tilt of the spectrum while the frequency scaling mainly treated the dissimilarities in spectrum resonance locations. The transforms were applied to a data base of average spectra of ten steady-state voiced-speech segments for seven male and six female speakers. Both the additive compensation and frequency warping, and particularly the integration of the two, was useful for both within- and across-sex normalization. The overall error rate in recognition experiments on the segment data base was reduced by 50% across sex, 60% within the female speaker group, and by 20% for the male speaker group. The best frequency warp paths were found to be nonlinear and strongly dependent on the speech segment. Parameters for a single, speaker-dependent, combined transform were also derived and the global transform was observed to be as effective as the equivalent speech-segment-dependent transform. The global transform parameters derived from as few as three segments yielded a performance level close to that attained with parameters extracted from the whole set of segments for one speaker. [Research supported by NSERC, Canada.]
Shakuhachi pitch and intonation: Application to computer music composition
Authors: Seltzer Linda A.
Published: January 2006

The music for the Japanese bamboo flute, shakuhachi, contains microtonality as well as continuous change of pitch in the course of long notes. Shakuhachi pitch and intonation may be studied by means of pitch detection analysis of the waveform. Or, alternatively, the pitch of a note may be described as the result of the following parameters: fingering, vocal tract shape, distance of the lips from the mouthpiece, and direction of breath. The melodic technique of the shakuhachi may be applied to computer music composition in the continuous frequency domain, without restriction to a scale consisting of discrete frequencies
Characterization of sea ice
Authors: Welsh James P.
Published: January 2006

Some definitions of icebergs and sea ice are presented. Various classification schemes for sea ice are discussed. The more common sea ice characteristics are presented with examples of measurement techniques and ranges of measured values. Consideration is given to potential for inference of acoustic properties from observed and measured sea ice characteristics. General overview is provided for applications of Arctic ice properties measurement experiments. These experiments include in-situ, airborne, and satellite measurements to obtain surface truth of remotely sensed ice characteristics particularly in the microwave frequencies.
Seismic events in the Arctic Ocean
Authors: Keenan Ruth E. , Dyer Ira
Published: January 2006

We present temporal and spectral characteristics of abyssally generated T phases observed in the Arctic Ocean. A large (∼ 1 km) horizontal hydrophone array suspended through the ice made it possible to verify the event as seismic. A simple propagation model accounts for the basic features of each event. We identified crustal and water wave arrivals? and the time difference between the longitudinal crustal wave and peak frequency arrival of the T phase established the range to the source. The intersection of the bearing of the T phase with the mid-Arctic ridge provided a corraborating range estimate. T-phase sonograms display characteristic pear shapes noted by others. The spectra have peak levels at 5 and 15 Hz and behave asymptotically as f4, f−4 above and below their peak level. The duration of the T phase is longer than can be consistently accounted for by any one mechanism. Possible mechanisms including source depth, multiple bounces, dispersion, and swarms are discussed. [Supported by SAI, Inc. and ONR.]
Icequakes in the Arctic Ocean
Authors: Stein Peter J. , Makris Nicholas C. , Dyer Ira
Published: January 2006

Resembling earthquakes, stress cracks in a floating ice sheet due to effects of winds, currents, and thermal gradients can excite various forms of elastic/acoustic wave propagation. The acoustic emission of several such icequakes were recorded during experiments on pack ice over the eastern Arctic abyssal plains. These events occurred in the near field of a 1-km aperture, 24-channel hydrophone array. The signals were bandpass filtered from 1 to 80 Hz. A recursive technique was used to locate these events and estimate the group speeds. Spectral characteristics are presented for different types of events. Interchannel correlation was performed to obtain distortion characteristics of the propagation path. Using these results, possible paths of propagation from source to receiver are discussed. [Work supported by ONR.]
Perception of musical interval tuning
Authors: Hall Donald E.
Published: January 2006

Pairs of tones have been generated under computer control and presented to musically trained subjects through headphones. Musical relevance is increased by using filtered pulse waves rather than sines. The stimulus intervals are harmonic (both tones heard simultaneously) rather than melodic. These intervals are mistuned in varying amounts in the vicinity of each traditional musical interval (e.g., around the 3:2 frequency ratio for the perfect fifth). Judgments of the relative degree of perceived mistuning are recorded and analyzed; although related to judgments of “dissonance,” these differ in significant ways. From the results we can infer relative ability to detect mistuning of different interval species. We find important differences from the traditional “consonance rank” ordering, and point out some implications for theories of keyboard tuning.
Analysis of intermodal coupling in piezoelectric ceramic rings
Authors: Benthien George W. , Martin Gordon E.
Published: January 2006

Many longitudinal vibrators employ a multi-element stack of piezoelectric ceramic rings as the active component. In order to design such transducers it is desirable to be able to measure longitudinal 33-parameters of piezoelectric ceramics using a single-ring geometry. Existing one-dimensional models are not adequate to describe the “longitudinal” mode for typical rings due to intermodal coupling. A piezoelectric finite element model was used to relate resonant frequencies to 33-ceramic parameters for certain selected ring geometries. However, since piezoelectric finite elements are not generally available, a simplified model was developed in which the necessary computations can be performed on a hand calculator. Guided by the finite element results, idealized mode shapes were used in conjunction with an energy method in order to obtain the simplified model. The resonant frequencies predicted by the simplified model were in excellent agreement with those obtained using the finite element method. This new model can be used to analyze a broad range of ring geometries. The authors have used the aforementioned models to obtain 33-piezoelectric parameters from measurements of resonant frequencies and low-frequency capacitance on ceramic rings.
Sound reflection from a fluid-loaded and masked elastic plate
Authors: Jackins P. D. , Gaunaurd G. C.
Published: January 2006

We have extended available computer codes that predict the classical reflections and transmissions of sound fields through sets of plane elastic layers [i.e., D. Folds and C. Loggins, J. Acoust. Soc. Am. 62, 1102–1115 (1977)]. The present layers separate possibly dissimilar fluids, are possibly viscoelastic, and are isonified from one side. The results we display include reflection (R) and transmission (T) coefficients as functions of frequency f and incidence angles θ. This computational tool has been used to predict the reflections from a single fluid-loaded elastic plate ideally covered with a lossy layer of homogeneous viscoelastic material, isonified from the coated side. Resonances in the coefficients are analyzed in the light of the resonance scattering theory (RST), which has not yet been applied to coated plates. The present model serves to assess the effectiveness of the masking layer to reduce the reflections from the metal plate. The predicted displays of calculations are presented for values of the material parameters in the ranges commonly found in practice. The lossy nature of the absorbing layer complicates the otherwise real parameters present in the analysis, and we quantitatively determine the shifts in resonance locations, the widening of the resonance peaks, and the overall reduction of the returned reflections caused by the masker's viscoelasticity.
Waveform synthesis—The Essex solution to cancelling periodic noise and vibration
Authors: Chaplin G. B. B.
Published: January 2006

The paper will give a basic description of the special method developed by the Essex Group for cancelling noise and vibration of a periodic nature. One of the advantages of the method is that the anti-noise or anti-vibration is produced by synthesis and does not depend on measurement of the original noise or vibration. The anti-noise and anti-vibration can therefore be injected at precisely the right time since no signal processing delay is involved. Examples will be given of applying the system to cancelling the noise at the exhaust of a diesel engine and the vibration at its mounting points, and to cancelling noise within a cab. The system is also highly selective and it is possible to cancel one source of noise whilst leaving other noise sources untouched. Experience on cancelling the exhaust stack noise on a cargo ship will be discussed.
Sample transport of levitated objects in a dual temperature resonance chamber
Authors: Robey J. , Trinh E.
Published: January 2006

Experimental studies show the feasibility of transporting an acoustically levitated object in an elongated resonance chamber filled with a gaseous medium, and having its opposite end zones at widely different temperatures (35° and 500 °C). Theoretical considerations describe mode behavior and the conditions for optimum power transfer into the heated region as well as the dynamic pressure distribution for various simple and complex modes. By properly introducing selected combinations of these modes, positioning and movement along the longitudinal axis and through the temperature gradient are possible. The reduced gravity of space would provide a useful environment for application of such a system in the areas of material processing, although the experimental work here has been carried out in the laboratory under 1 g with low-density materials.
Test of the nearfield holography technique using an unbaffled, uniformly oscillating disk
Authors: Beyer Todd B. , Maynard Julian D.
Published: January 2006

Experimental results of nearfield holography measurements of the radiation from an unbaffled, uniformly oscillating disk have been compared with the exact theoretical solution. The experimental source is a disk 1 m in diameter constructed of an aluminum honeycomb material that is lightweight and structurally rigid. Eight drivers located beneath the disk can be individually adjusted so that the disk's surface vibrates inphase with constant amplitude. The radiation efficiency determined from the measurements at two frequencies (150 Hz, 250 Hz) agrees closely with the radiation efficiency calculated from the exact theory. The theoretical solution makes use of oblate spheroidal functions to compute the pressure on the surface of an ideal rigid disk vibrating with uniform normal velocity. [Work supported by the National Aeronautics and Space Administration.]
Edgetone spectra
Authors: Nolle A. W.
Published: January 2006

It is well known that as the velocity of an air jet impinging on an edge is increased, various regimes of periodic oscillation are found in which the frequency increases with jet velocity. It will be shown that also there are narrow ranges in which the signal consists of two or more components, not harmonically related, and various intermodulation products. Ultimately, as the jet velocity increases, the signal becomes a noise band. Extensive data on these phenomena will be shown. In the present work the jet-to-edge distance is 4–12 mm, and the jet velocity lies between 5 and 30 m/s. In resemblance to a common practice in organ pipe construction, the piece whose edge is struck by the jet is unsymmetrical, consisting of a 45° wedge mounted so that one side is parallel to the center line of the jet.
The conical bore in musical acoustics
Authors: Ayers R. Dean , Eliason Lowell J. , Mahgerefteh Daniel
Published: January 2006

A review of textbooks and research literature reveals that there is room for improvement in the treatment of this simple but important bore shape. We use plots of acoustic pressure standing waves to show students in a descriptive course that a complete cone and an open pipe of the same length have the same natural frequencies. These plots also predict the qualitative behavior for the frequencies of a frustum closed at the small end and suggest a rigorous approach to their calculation. Incorporating the open end correction for a straight pipe plus a correction for viscosity and thermal conductivity in the calculation results in agreement with experimental values to within 0.5%. In order to understand the inharmonicity of the frustum's frequencies from a different approach, we have calculated the pressure impulse response for reflection from its closed end. We find that this consists of a delta function plus an inverted, exponentially decaying wake. Experimental observations confirm both that shape for the impulse response and its contribution to the evolution of the waveform through several reflections.
Reverberation model of the arctic channel in the presence of ridge scattering
Authors: Jacyna Garry M. , Friedman Alan I.
Published: January 2006

A ray-theoretic model is advanced to describe the average reverberation intensities within an Arctic channel bounded by a pack-ice surface and an upward-refracting sound-speed profile in the presence of pressure ridge scattering. The surface is assumed random and a phenomenological model based on Kirchoff assumptions in the high-frequency limit is derived. Relevant parameters are estimated from data supplied by Chapman and Scott using a curve-fit routine. Ridges are modeled as inverted cones, the surfaces of which are also assumed random. This simplifying geometric assumption allows for a highly tractable technique. Forward as well as backscattered geometries are treated with no assumptions regarding source/receiver range or depth and ridge location or draft. Applications to sonar receiver performance will be discussed in a companion paper. [Work supported by ONR.]
Speaker-independent isolated word recognition for small-size hardware using multiple valued Walsh Mel-cepstrum
Authors: Watari M.
Published: January 2006

In this paper, the multiple-valued Walsh Mel-cepstrum is proposed as a speech parameter for a small-size speech recognizer. The new speech analysis is based upon the multiple-valued Walsh transform instead of the Fourier transform used in the standard Mel-cepstrum. This makes it possible to obtain good speech parameters by using a small microcomputer. The analysis process has four stages: the multiple valued Walsh transform, the log magnitude computation, the Mel-scale conversion, and the inverse multiple valued Walsh transform. The multiple-valued Walsh transform operator uses multiple-valued coefficients (p/2n + q/2nj) corresponding to ejw used in the Fourier transform. This operation requires only additions, subtractions, and shifts. Speaker-independent isolated digit recognition experiments were carried out to evaluate the new speech analysis method. The recognition error rates for Walsh Mel-cepstrum, multiple-valued Walsh Mel-cepstrum (16 valued), and Mel-cepstrum were 6.4%, 3.2%, and 3.1%, respectively.
Increased low-frequency sensitivity in the ears of Kauai moths: Response to the acoustic signals of the Hawaiian Hoary Bat
Authors: Fullard James H. , Bellwood Jacqueline J.
Published: January 2006

The auditory sensitivity of tympanate moths from the island of Kauai was studied to examine the sensory effects placed on these insects by the site's sole bat species, Lasiurus cinereus semotus (Hawaiian Hoary Bat). Audiograms were neurologically determined and then subdivided into four bandwidths to render an estimation of sensitivity (audiogram area) at 5–25, 30–50, 55–75, and 80–110 kHz. Compared to moths analyzed from sites in Canada, Africa, and Papua New Guinea, Hawaiian moths reveal a reduced high-frequency (> 50 kHz) sensitivity coupled with a significantly higher sensitivity to frequencies below 30 kHz. Tuning is very broad with best frequencies extending from 9 to 33 kHz for various species. Free-flying individual bats were recorded as they haunted at the same site where moths were captured. L. cinereus emits two distinct signals as it flies: a typical FM, harmonically complemented, echolocation call centered at 27.9 kHz and another FM signal of 9.8 kHz, commonly emitted during agonistic encounters with other bats while foraging. We suggest that this bat gleans echolocation information from its low-frequency social call and thereby provides another set of acoustic frequencies to which sympatric moths have become sensitive. [Work supported by National Geographic Society and NSERC.]
Responses of cerebellar units in FM and CF-FM bats to acoustic stimuli
Authors: Jen Philip H.-S. , Kamada Tsutomu , Sun Xinde
Published: January 2006

With pure tone pulses (35- and 0.5-ms rise-decay times), a total of 499 units from FM bats (Myotis lucifugus and Eptescius fuscus) and 401 units from the CF-FM bats (Pteronotus parnellii parnellii and Pteronotus parnellii rubiginosus) were isolated from a large area of the, cerebellar vermis and hemispheres. Response latencies ranged between 1.2 and 68 ms but most were below 10 ms. Phasic units, phasic bursters, and tonic units were observed. Threshold curves of these units were either narrow, broad, or irregular. In the CF-FM bats, units with BFs between 60 and 63 kHz corresponding to the predominant CF frequency were sharply tuned and showed off responses upon cessation of the stimulus. In both bats, variation of BFs of isolated units from penetrations within the same lobe was smaller than those from different lobes. A study of frequency tuning and representation suggests that a bat's cerebellum can effectively process the predominant components of biologically significant signals. Responses to a 4 ms upward and downward sweeping FM stimuli and pure tone pulses were studied in 70 units of the Eptesicus fuscus. Most units were more sensitive to a downward sweeping FM stimulus than to a pure tone pulse. An upward sweeping FM stimulus was as effective as a 35-ms pure tone pulse. A study of directional sensitivity of 47 units showed that 33 units were most sensitive to a sound delivered from the front. The other 14 units were most sensitive to a sound delivered from 20°–30° lateral. [Work supported by NSF 80 07348 and USPH 1-K04-NS-00433 to P. Jen.]
Frequency discrimination of tones presented either inside or outside of “notches” in band-reject noise
Authors: Emmerich David S. , Fantini Deborah A. , Brown William S.
Published: January 2006

At the 104th meeting of this Society we reported data consistent with the notion that, in frequency discrimination, information is integrated over a wide frequency range. These data were obtained from subjects who were asked to discriminate tones which were presented in the centers of “notches” of various widths in backgrounds of band-reject noise. The present research extends the earlier findings by presenting the tones close to the edges of bands of rejected frequencies in an attempt to determine the relative importance of activity from frequency regions higher and lower than the nominal frequencies of the tones being discriminated. In addition, control experiments were conducted in which the tones were presented in the passbands of band-reject noise rather than inside the notches. Again the results suggest that activity remote from the peak of the distribution of activity created by the presentation of a tone is important in frequency discrimination. A reasonable interpretation of the data is that remote effects extend more widely on the high-frequency side than on the low-frequency side of the tones to be discriminated. This interpretation is somewhat clouded by the unexpected finding that in some conditions adding to the spectral content of the noise background improved rather than impaired performance.
On the form of the masking function for intensity discrimination of pure-tones
Authors: Viemeister Neal F. , Bacon Sid P.
Published: January 2006

The published data on intensity discrimination of tones does not permit detailed characterization of how the difference threshold (ΔI) varies with intensity (I) over a wide intensity range. In the present study, masking functions (ΔI vs I) were obtained for 200-ms, 1-kHz tone bursts over a 115-dB range with I varied in 2.5- or 5-dB steps. For intensities between 5-dB SL and 95-dB SPL, the masking functions were in reasonable agreement with recent reports: the Weber fraction, ΔI/I, was approximately constant up to 40 dB SPL and then decreased monotonically up to 95 dB SPL. For I near absolute threshold, some subjects showed a slight “sensitization,” in which ΔI was below absolute threshold. Above 100 dB SPL, the Weber fraction increased monotonically with increasing intensity, suggesting mild “saturation.” These data should permit more refined modeling of intensity discrimination of tones and of certain complex waveforms. [Supported by NINCDS NS12125.]
Forward masking tuning curves obtained in the presence of a backward noise masker
Authors: Shelton Brian R. , Booth John C.
Published: January 2006

The combined effect of a backward and forward mask has often been noted to be difficult to predict on the basis of the masking obtained from either masker alone. This report describes the nonlinear effects of a backward mask on the shape of forward masking tuning curves. Five-point psychophysical tuning curves were obtained in two observers, using 200-ms forward maskers of 500, 750, 1000, 1250, and 1500 Hz, and a 10-ms, 1000-Hz signal presented at 40 dB SPL. The forward mask had a 2-ms rise/fall time and the signal was fast-gated 3 ms after the forward mask. The backward mask was a 100-ms burst of white noise, presented 3 ms after the signal offset, with a fast rise/fall. Tuning curves were obtained in quiet, and with backward maskers of 20, 50,and 80 dB SPL. Tuning curves became sharper as the level of the backward mask was increased. The tip of the tuning curve dropped by at least 20 dB in the presence of the 80-dB backward mask, and by less than 10 dB at other frequencies. The results are discussed in terms of the strategies employed by listeners to detect a signal in the presence of a forward mask. [Work supported by NSERC.]
Improved 1-bark bandwidth auditory filters
Authors: Sekey Andrew , Hanson Brian A.
Published: January 2006

A critical band filtering function attributed to Schroeder [A. J. Fourcin, Rapporteur, “Speech processing by man and machine,” in Theodore H. Bullock (1977), Recognition of Complex Acoustic Signals, Dahlem Konferenzen, Berlin] is modified so as to (i) ensure 1-bark bandwidth, (ii) have asymptotic slopes which reflect masking characteristics of the human ear, and (iii) attains its maximum at 0 bark. A bank of the new filters spaced at 1-bark intervals is presented, and its smoothing effect on a vowel spectrum is illustrated.
Discrimination of noise as a function of bandwidth and duration
Authors: Hanna Thomas E.
Published: January 2006

A same-different task was used to study the discriminability of noise bursts as a function of duration (0.1, 0.4, 1.6, 6.4, 25.6, 102.4, or 409.6 ms) and bandwidth (50, 200, 800, or 3200 Hz) of the burst. Each trial contained either identical (same) or independent (different) samples of noise, with the noise bursts sampled randomly for each trial. Performance improved as a function of bandwidth, as would be expected based on the increase in information in the stimulus. However, performance changed nonmonotonically as a function of duration, even though there is a similar increase in information as duration increases. For each of the four bandwidths, performance improved up to a duration of 25.6 ms, but declined with further increases in duration. Subsequent experiments were conducted to explain the decrease in performance for longer durations. The results will be discussed in terms of the peripheral and central determinants of performance for this task. [Work supported by NIH.]
Acoustic signal processing: An interdisciplinary field
Authors: Bartram James F.
Published: January 2006

This paper presents an overview of the field, from the perspective of the recent Associate Editor for Acoustic Signal Processing, Journal of the Acoustical Society of America (JASA). The subject matter is defined, both as to what it is not as well as to what it is, and the history of the Journal category is briefly presented. The author stresses the fact that signal processing cuts across all nine of the Society's technical areas and many other nonacoustical scientific disciplines. To emphasize the interdisciplinary character of signal processing, an examination of the subject matter contained in the various signal processing papers published in JASA and other journals in the past three years is presented.
Nearfield acoustical holographic techniques used to visualize radiated sound fields
Authors: Maynard J. D.
Published: January 2006

The term signal processing is customarily used in reference to the analysis of data obtained as a function of time; similarly the term image processing refers to the enhancement of two-dimensional spatial data. For wave fields, such as optical or acoustical fields, the wave equation couples the space and time variables and permits an extremely powerful signal processing technique referred to here as generalized holography. In this technique temporal data are measured over a two-dimensional spatial surface, and the data are processed to reconstruct and visualize the entire three-dimensional wave field. The power of this technique arises from the enormous expansion of information which occurs when going from the two-dimensional measurement to the three-dimensional reconstruction. Although the theory of generalized holography is exact, conventional optical and acoustical holography suffers from serious intrinsic limitations, such as the spatial resolution being limited to the radiated wavelength. At The Pennsylvania State University we are developing a new acoustic generalized holographic technique, called Nearfield Holography, which overcomes the limitations of conventional holography. In this technique a brief measurement with a microphone array can be used to visualize the surface motion of a complex vibrating structure, to reconstruct the entire three-dimensional sound pressure field, particle velocity field, and vector intensity field, and to map the flow of acoustic energy from the sources to the farfield. With this system one may pinpoint (within centimeters) acoustic energy producing sources on vibrators with small structural features (such as musical instruments) even when the radiated wavelength is several meters. In this talk our most recent Nearfield Holography equipment and processing algorithms will be described, and computer graphic reconstructions of vibrating surfaces (moving in real-time) and radiated sound fields will be shown. [Nearfield holography research is supported by the Office of Naval Research (Physics Program) and the National Aeronautics and Space Administration.]
The evaluation of the effectiveness of three hearing protection devices at an industrial facility with a daily TWA of approximately 107 dB
Authors: Royster Larry H. , Royster Julia Doswell
Published: January 2006

Three hearing protection devices (Norton Sigma Comfit earplug, Safeco Silenta earmuff, and E-A-R foam earplug) were compared for their effectiveness in preventing both daily temporary threshold shifts (TTS) and shifts between sequential annual audiograms (%BW statistic) [L. H. Royster and J. D. Royster, in Personal Hearing Protection in Industry (Raven, New York, 1982)]. The hearing protection devices varied significantly in preventing TTS. Comfit wearers exhibited the largest TTS shifts, which differed from zero at 500–4000 Hz. Significant TTS was found for Silenta wearers at 500 Hz, and for E-A-R wearers at 1000–2000 Hz. E-A-R wearers showed nonsignificant improvements at 3000–6000 Hz, and Silenta wearers showed significant improvement at 6000 Hz. Employees' previous three annual audiograms were evaluated to compare %BW values for the wearer groups. These values were 53% for Comfit wearers, 45% for Silenta wearers, and 26% for E-A-R wearers. Previous research indicates that the %BW statistic for a protected or non-noise population with past audiometric test experience will be less than 30%.
A new approach to auditory depth perception
Authors: Gatehouse R. W.
Published: January 2006

Acoustic depth judgment studies have generally used static position sources, and asked subjects to estimate the distance of one source only, or one source relative to a second. Here, blindfolded subjects were asked to “align” a moveable medial sagittal plane comparison source with a static 1.O-kHz standard, 3 m away from them. The comparison source's starting position could be at 0 (position of standard), ± 100 or ± 200 cm away and was either a 0.7- or 1.3-kHz narrow-banded tone. The two sources alternated in 3-s bursts till the Sindicated “alignment” was achieved. Half the subjects received the 1.0 and 0.7 kHz pairing first then the 1.0 and 1.3 kHz pairing and vice versa. Three trials were given for each pairing from each starting position. The mean error scores (i.e., mean differences between “true” and estimated source position) were analyzed. The results indicate that observers overestimate the distance of the standard when initial position of the comparison is close to them, and underestimate it when the comparison starts from beyond the “0.” Comparison frequencies performances did not differ. The results partially confirm Mershon and Bowers [Percept. 8, 311–322 (1975)] using a more standard depth perception paradigm.
Effects of signal starting phase on diotic and dichotic detection in reproducible noise masker
Authors: Gilkey R. H. , Kollmeier B.
Published: January 2006

Dolan, Hirsh, and Yost [J. Acoust. Soc. Am. 70, 886–887 (1981)] and Gilkey, Hanna, and Robinson [J. Acoust. Soc. Am. Suppl. 1 68, S59 (1980)] have shown that the detectability of a tonal signal varies greatly as a function of the phase relationship between the signal and a reproducible noise masker under diotic (NoSo) presentation. However, detectability varies only slightly, if at all, as a function of phase under dichotic (NoSπ) conditions. In the present experiment an adaptive two-alternative forced-choice procedure is used to obtain detection thresholds for a 500-Hz tonal signal in the presence of each of six individual samples of reproducible noise. Six binaural conditions (NoSo, NuSo, NπSm, NoSm, NπSo, and NoSπ) were investigated in combination with six signal starting phases (0°, 60°, 120°, 180°, 240°, and 300°). The results indicate that although the magnitude of the phase effect decreases with larger masking level differences, it is similar in form for all conditions with the same masker. [Work supported by NINCDS Grant NS 03856, and the Fulbright Commission, Federal Republic of Germany.]
A comparison of the temporal response of the auditory system under dichotic and diotic conditions
Authors: Kollmeier B. , Gilkey R. H.
Published: January 2006

Most of the classical experiments on binaural masking have held the interaural phase of the noise, as well as that of the signal, constant throughout the observation interval. In this experiment the detectability of a 500-Hz tonal signal is investigated in the presence of a 2000-Hz low-pass noise, which is switched from Nπ to No at the midpoint of the observation interval. For a 20-ms Sπ signal there is approximately a 15 dB increase in detectability when it is presented late in the interval, relative to early in the interval. The time course of the change in detectability can be used as an indication the decay of activity in the binaural system. This decay of activity is compared to a similar diotic configuration in which the level of the noise is decreased by 15 dB. The results indicate that the decay of activity in the binaural system is more “sluggish” than the decay of activity in the monaural system. [Work supported by NINCDS Grant NS. 03856, and the Fulbright Commission, Federal Republic of Germany.]
Response durations and false alarm reaction times
Authors: Orr John L.
Published: January 2006

Three Long-Evans rats were initially trained to depress and hold down a response lever in the presence wideband white noise pulsing at 6.7 Hz for a variable period between 4 and 6 s to obtain a 45-mg food pellet. This reinforcement schedule is the same as operating a one-lever press-release psychophysical procedure with a variable foreperiod of 2–4 s and a 2-s trial duration with only no-signal trials. The initial training of the nonsignal trial correct response of holding the lever down throughout the trial permits the measurement of the temporal characteristics of the background level of release responses in the absence of a history of contingent reinforcement for releasing the lever. After training to asymtotic performance on the nonsignal trials alone, 50% signal trials (where the noise. pulsation rate changed to 12.5 Hz) were included. Discrimination training was followed by changes in response duration histograms and false alarm reaction time histograms. In two of the three rats, there was an increase in the probability of incorrect lever releases 3–6 s after the lever was depressed. [Work supported by NIEHS grant RO1ES02750 to R. M. Lebovitz.]
Digital signal processing for sound and vibration analysis: A tutorial
Authors: Burgess John C.
Published: January 2006

The literature on digital signal processing is based on four kinds of transform: Fourier transform (FT), z-transform, Fourier series, and discrete Fourier transform (DFT). Digital implementation uses only the DFT, usually in the form of an FFT algorithm. Some popular texts attempt to describe digital results by using analog (usually FT) analysis. Some have presented analytical results for infinite length random signals and then used illustrations based on finite length periodic signals. What “windows” do and their different effects on random and periodic signals seldom are clearly identified. There are two implementations of the DFT that differ by a constant factor. Fast convolution and correlation analyses require special handling to avoid “circular” effects. The DFT treats each finite length sample from a random signal as if it were a sample from a deterministic signal. The purpose of this tutorial is to illuminate these and other potentially troublesome aspects of digital signal processing pertinent to applications in sound and vibration analysis.
Noise source identification by coherence and acoustical methods using FFT analyzers
Authors: Hodgson Thomas H.
Published: January 2006

The practical understanding of the two-channel FFT spectrum analyzer as used in noise source identification will be addressed. In particular, the measurement of the coherence function and the acoustic intensity, the latter by both two-microphone and surface methods, will be described. Problems related to amplitude and phase information and the design of most analyzers as they relate to Fourier transform theory or electrical network practice will be discussed. Several case studies of noise source identification measurement will be presented in order to demonstrate some of the points related to FFT analyzer use described in the paper. Both steady-state and impact noise problems will be considered with reference to causes of possible errors.
Effects of cochlear high-frequency hearing loss on BSER
Authors: Shepard Neil T. , Webster John C.
Published: January 2006

The extensive use of Brainstem Evoked Response Testing (BSER) for the detection of mass lesions involving the VIIIth nerve necessitates knowledge of the effects of hearing loss of cochlear origin on BSER. Thirty-one ears displaying varying degrees of sensorineural, high-frequency (beginning above 500 Hz) loss of sensitivity of cochlear origin were tested with BSER. A 100-μs rectangular pulse was used to excite TDH-39 earphones at a rate of 11.3 pulses/s. Utilizing monaural stimulation at two to five intensities per ear, responses were obtained from a vertex to ipsilateral mastoid electrode montage. Linear regression analysis lines for waves I, III, and V latencies versus average degree of loss for various combinations of audiometric test frequencies are shown. The dependence of BSER wave latencies on individual and audiometric test frequencies and various combinations at and above 1 kHz are shown via the use of correlation coefficients. The results compare well with earlier results [A. C. Coats and J. L. Martin, Arch. Otolaryngol. 103, 605–622 (1977)] and imply the dependence of results on click stimulus parameters and earphone spectral characteristics. Also suggestions for normative limits of wave V latency shift for varying degrees of high-frequency cochlear hearing loss are given.