bluuebunny/crossref_metadata_2025 huggingface dataset:
Cost effectiveness of construction noise abatement
Authors: Kessler F. M. , Schomer P. D.
Published: October 2005

The U. S. Army Corps of Engineers Construction Engineering Research Laboratory has sponsored research to determine the cost benefit relationships for Army housing construction noise abatement. Before these cost benefit relationships could be obtained, realistic analytical models of construction site noise emissions were developed. The models were compared with noise measurements of construction activity at Fort Hood, Texas. Data comparing the construction noise model with noise measurements for different phases of construction are presented and discussed. The construction noise models are used together with proposed acceptability criteria to quantify the degree of noise abatement required at Fort Hood, Texas. Data were obtained and are presented for the costs of various construction equipment and site noise abatement methods required to achieve property-line or noise-sensitive land-use criteria sound levels. Preliminary noise-abatement cost benefit relationships are presented.
Perceptual consequences of array design
Authors: Wightman F. L. , Queen D.
Published: October 2005

Configuration of broadband loudspeaker elements into arrays impacts phase and amplitude characteristics versus frequency at different listening positions. It can also affect the perceived arrival times versus frequency when early-room reflections reach the listening position. The perceptual consequences of such variations are examined utilizing previously reported experimental procedures for evaluating localization and image recognition.
Basic operation of the Barcus—Berry AudioPlate
Authors: Ayers R. Dean , Kellam John M.
Published: October 2005

The Barcus-Berry AudioPlate is a new type of speaker which employs a large piezoelectric element to drive a glass plate in bending motion. Tests conducted in an anechoic chamber indicate a useful frequency range extending from about 2 kHz to well above 20 kHz. Particularly prominent above 10 kHz is an unusual maximum in the radiation pattern in the shape of a cone, with its half-angle decreasing as the frequency is raised. The maximum and its behavior are explained quite simply as a diffraction effect from the waves propagating in the plate. Tests in which an unmounted plate has been driven in resonance for particularly simple modes have produced radiation patterns that can be compared with the results of a straightforward analysis. Calculations have confirmed some interesting details of these patterns.
Audio teleconference rooms—general guidelines
Authors: Stockbridge C. D. , Curtis T. H.
Published: October 2005

General guidelines for the installation of audio teleconference equipment have been written and distributed to Bell System Operating Companies. The guidelines cover conference-room acoustics, teleconference equipment, suggested equipment arrangements, and transmission considerations. The guidelines recommend that microphones be installed in a conference situation so as to provide (1) a 20-dB (speech) signal-to-(ambient room)-noise ratio and (2) a 6-dB (direct speech) signal-to-(reverberent speech)-signal ratio. As a rule of thumb, microphones placed on a table within two to three feet of a talker give good results, microphones mounted in ceilings do not. Charts, tables, and worksheets are included which enable an installation engineer to determine the appropriate microphone-to-talker distance.
Development of a vehicular directional siren with variable beamwidth utilizing a broadside loudspeaker array
Authors: Fisher R. L. , Toth D. D.
Published: October 2005

A directional electronic siren was developed which could be electrically switched from a narrow beam in the forward direction for open-highway usage to a broader beam for use near-roadway intersections. The beam was produced by a vehicular-roof-mounted broadside array consisting of four compact commercial electronic-siren loudspeakers spaced 15.2 cm (6 in.) apart. Two electronic systems were developed to broaden the inherently narrow beam of the broadside array. One system swept the beam from side to side in 1.4° increments. Another used a filter network to broaden the beamwidth by effectively changing the broadside array to an endfire array with asymmetrical phasing of the loudspeakers. A-weighted sound-pressure-level measurements made in an anechoic chamber and outdoors on a vehicle show that the maximum forward sound pressure level of the directional siren was 8–10 dB higher than a single commercial electronic siren, and that the desired directionality properties were achieved.
Computer graphics for speech research and for communication
Authors: Denes Peter
Published: October 2005

Computer graphics has become a widely used, convenient tool for speech studies. Its speed has made it particularly effective for improving man—machine interaction in real-time computing (and information retrieval) situations: examples of such applications will be given. Despite its widespread use, not enough is known about the factors that make data displays more informative, easter to understand, and visually more interesting. As an example, the speech spectrograph—that widely used speech research tool—and various unsuccessful attempts to make its output easier to read will be described. In these experiments the conventional gray-scale method for representing spectral intensity has been replaced by stereography, or by perspective, or by color: possible reasons for the lack of acceptance of these alternative methods will be discussed. Finally, there is one further important reason why a better understanding of graphics is needed. It is expected that inexpensive graphic displays as well as speech synthesizers will become available for man—machine interaction soon. A good understanding is therefore needed of which interactive situations are served better by graphics, by voice, or by a combination of the two.
Interactive laboratory system on an IBM 370/158
Authors: Purcell Edward T.
Published: October 2005

A broad, flexible, and powerful set of interactive programs for the acoustic analysis of speech have been developed at the Speech Communications Research Laboratory of Santa Barbara. This interactive laboratory system (ILS) was originally developed for use on minicomputers. The present paper will describe an adaptation of ILS for use within a major time-sharing configuration. Operating on an IBM 370/158 under OS, ILS remains an extremely powerful tool for acoustic phonetic research. An added benefit of operating in this environment is the availability of other resources which are available at our installation, e.g., SPEAKEZ, SPSS, SAS, MINITAB, etc., which are extremely valuable for rapid data reduction.
Sound propagation in the near surface of a spherical shell
Authors: Wong E. H.
Published: October 2005

An elastic spherical shell was insonified by a point source. A two-dimensional cross section of the sound field near the shell in both the direct field and shadow zone is represented graphically in three dimensions with relief representing amplitude. The results are presented in a motion picture filmed from a cathode ray tube of an interactive computer graphics terminal. A visualization of where the sound source is located, and the distribution of the sound field in the neighborhood of the shell is presented. The movie allows us to compare theoretical and experimental directivity patterns and to dynamically view the changes in the directivity pattern while sweeping through a range of frequencies near ka = 20.
Acoustic diffraction graphics
Authors: Kendig R. P. , Simowitz K. T.
Published: October 2005

The farfield diffraction and backscattered pressure fields from an impedance covered half-plane are investigated with the aid of an interactive computer graphics system. Color and three-dimensional imagery are employed in a movie sequence to illustrate the influence of the impedance cover. The movie demonstrates that the impedance cover on the insonified surface dominates the behavior of diffracted and backscattered pressure in the half-space into which it faces.
Effects of selective adaptation on the perception of voicing contrasts in Thai
Authors: Donald S. L.
Published: October 2005

A series of four adaptation experiments using Thai subjects is being carried out in order to investigate a possible phonetic contribution to adaptation effects on a voicing continuum. Thai subjects divide a VOT continuum into three phonological categories at the labial and dental places of articulation but only two phonological categories (voiceless inaspirates and voiceless aspirates) at the velar place. The effects of −70-msec labial and velar adaptors on both a labial and a velar VOT continuum were determined. If there is a phonetic contribution to adaptation, a −70-msec velar adaptor should produce a significant boundary shift between voiceless inaspirates and voiceless aspirates on both the velar and the labial VOT continua, while a −70-msec labial adaptor will affect neither boundary. If, however, adaptation takes place solely at an acoustic level, both adaptors will shift the boundary between velar voiceless inaspirates and voiceless aspirates, but will not affect the boundary between labial voiceless inaspirates and voiceless aspirates. Preliminary results indicate the presence of a phonetic contribution to the adaptation effect. [Work supported by NICHD.]
Further observations on the function relating the voicing boundary to changes along a place continuum
Authors: Repp Bruno H.
Published: October 2005

At the last meeting, I presented data showing that the function relating the voicing boundary for stop consonants preceding /a/ to changes along a place continuum (F2 and F3 transitions) was generally increasing but had an unexpected peak around the labial-alveolar boundary, where the F2 transitions were flattest. The data were from a single subject—myself. In the meantime, data from several other listeners have obtained, revealing some striking individual differences in the overall shape of the function, but also some agreement in its local variations. Three new series of stimuli were constructed with different steady-state frequencies of F2 but identical F2 onset frequencies. Their purpose was to shift the point where the F2 transitions become flat up and down the place continuum, in order to determine whether the peak in the voicing boundary function would shift, too. Preliminary data suggest that peaks in the function are not related to flatness of F2 transitions. Data from several listeners will be reported.
On measuring the DL for pitch memory with the transformed up-down psychophysical method
Authors: Smiarowski Richard A.
Published: October 2005

Traditionally, the difference limen for pitch memory has been determined by generating a psychometric function with the method of constant stimulus differences. This paper discusses the application of a more efficient psychophysical technique to the measurement of the pitch DL. In this experiment, the listeners were instructed to identify which of two tones in a two-alternative forced-choice paradigm was higher in pitch. Threshold was defined as that frequency difference necessary for 79.4% correct discrimination. This performance level was determined with the transformed up-down strategy [Levitt, J. Acoust. Soc. Am. 49, 467–477 (1971)]. During the up-down testing, the variable frequency was increased and decreased in equal ratio fashion by means of a programmable frequency synthesizer capable of providing six-digit resolution. The pitch DL's generated with this method were comparable to those measured with the more cumbersome method of constants [Harris, J. Acoust. Soc. Am. 40, 43–46 (1966)]. [Supported by V. A. Medical Research Project 7990-01.]
Envelope detection as a source of monaural phase effects
Authors: Buunen T. J. F.
Published: October 2005

A change in the phase relations in a three-component signal is audible if certain conditions of amplitudes and frequencies are met. These conditions have been expressed as an audibility region and they show that combination tones are an important source of phase effects. However, for low periodicities of the envelope of the signal (e.g., 20 Hz) the envelope will also be detected. This is relevant for phase effects because phase changes affect the envelope. Experiments were performed to estimate the range of periodicities for which envelope changes can be heard. Observers were asked to discriminate between two signals each consisting of S(inusoidally) A(mplitude) M(odulated) noise added to a SAM tone. The modulation frequencies of the noise and the tone were equal; in one signal they were in phase, in the other in antiphase. If any difference between the two signals was audible it was assumed that the envelopes of both tone and noise were detected. By changing the modulation frequencies an estimate of the working area for envelope detection was obtained. Comparison with the audibility region of phase changes shows that envelope detection is responsible for only a small part of this region.
Low-frequency hearing loss: perception of filtered speech, psychophysical tuning curves, and masking
Authors: Thornton Aaron R. , Abbas Paul J.
Published: October 2005

Two subjects with low-frequency hearing loss were evaluated to determine whether their responses to low-frequency stimulation might be a result of stimulation of nerve fibers with higher characteristic frequencies. One subject showed large low-frequency threshold shifts in the presence of high-pass noise or a 2000-Hz tonal masker. Psychophysical tuning curves for 500- and 800-Hz probe signals were peaked above 2000 Hz. This subject was also tested with high-pass, low-pass, and unfiltered speech both in quiet and in the presence of a high-pass noise masker. Results were interpreted as showing relatively little encoding of low-frequency speech by high-frequency nerve fibers. The second subject had masking patterns and psychophysical tuning curves which were most consistent with detection of low-frequency signals by nerve fibers with low characteristic frequencies. Psychophysical tuning curves of both subjects were compared to those obtained from subjects with high-frequency hearing loss. Implications for the diagnosis of low-frequency hearing loss and the use of hearing aids are discussed. [Work supported by NIH.]
Effect of amplitude envelope on the pitch of sine-wave tones
Authors: Hartmann W. M.
Published: October 2005

In a 2 IFC constant-stimuli up-down experiment subjects compared the pitches of sine-wave tones with 20-msec rectangular envelope with the pitches of sine-wave tones with exponentially decaying envelope (60-dB decay time = 60 msec). In a survey experiment the exponential envelope raised the pitch of a tone compared to that with a rectangular envelope for 13 out ot 14 subjects. For those subjects exhibiting the positive shift the mean pitch shift Δf was 11 Hz ± 4 for sine waves with frequency f between 800 and 850 Hz. A detailed parametric study with four subjects revealed the following: (1) The pitch shift is not simply an overall loudness effect. It is present in much the same form for envelopes of varying relative loudness. (2) The relative pitch shift Δf/f decreases with increasing f2.6% at f=412, 1.7% at f=825, 1.0% at f=1650, and 0.7% at f= 3300. [Work supported by the NSF.]
Pitch of two-tone complexes
Authors: Feth L. L. , Ramsey J. , O'Malley H.
Published: October 2005

Helmholtz reported that the pitch of a two component complex tone could be shifted by slightly increasing the amplitude of one component. The pitch was shifted toward the frequency of the more intense component. Helmholtz attributed the pitch shift to a variation in the instantaneous frequency of the two tone complex. This laboratory has previously investigated the discriminability of pairs of two component complex tones with identical small amplitude differences between the components of a pair. Discriminability seemed to depend upon the difference in their envelope-weighted instantaneous frequency fluctuations. In the study to be reported, we have asked listeners to match a two component, nonfluctuating complex tone to each member of pairs of the two component complex tones described above. The differences in pitch so obtained will be compared with the differences in envelope-weighted instantaneous frequency fluctuations to relate the pitch of each fluctuating two tone complex to its averaged instantaneous frequency. [Work supported by a grant from the National Institutes of Health.]
Technique for observing loudspeaker wave-front propagation
Authors: Nemoto I. , Iwahara M. , Onoye H.
Published: October 2005

Conventionally the measurement of sound field has been restricted to steady state measurements with sine waves used as inputs. However, a new method is developed utilizing a computer graphic. Scanning a microphone in front of a loudspeaker, the impulse responses of a loudspeaker are measured at 3000 points and are stored in the computer. The data are in turn processed and converted to the patterns of sound pressure distribution on a cathode ray tube at 10-μsec (100-kHz) intervals. The resulting display is photographed sequentially for observation of wave-front propagation from the loud speaker. Projecting this film at a proper speed simulates in vision a sound propagation from the speaker including interferences such as reflection and diffractions in the space in front of the speaker. This method is applied to measure different types of transducers which includes: diaphragm/horn, ribbon/horn, paper cone, hard-dome, soft-dome, and an air-motion transformer type. Behaviors of a two-way system including its frequency divider are also observed. The results are shown in the 16-mm movie film.
Cytochemical bioassay of ACTH in noise stressed subjects
Authors: Brister N. W.
Published: October 2005

The potential value of a newly developed hormonal bioassay for ACTH was investigated with respect to its usefulness for diagnosing noise sensitive subjects. Preliminary results of the effects of noise stress (100 dB, 60–8000 Hz, × 15–60 min) on circulating levels of ACTH were examined in laboratory animals and human subjects. Data indicate that immediately after onset of noise stress there is a transient adrenocortical activation, followed by a marked suppression of the pituitary-adrenal axis which occurs within one hour after onset of noise so that circulating ACTH levels fall to extremely low levels.
Hormonal aspects of noise stress: relationship of blood steroid levels to RNA synthesis in adrenal cortical cells and hippocampal neurons
Authors: Durkot Michael John
Published: October 2005

Male Rats exposed to a 60-min interval of noise (100 dB re 20 μN/m2, 0.3–20 kHz) exhibited a biphasic increase in circulating corticosterone at 0 and 12 hours after cessation of noise stimulation. Adrenal fasciculate cell RNA levels followed a similar temporal pattern of changes whereas an inverse pattern of neuronal RNA changes were evidenced in the hippocampus. The data indicate that noise induced adrenocortical activation appears to be associated with inhibition of ribosomal activity in the limbic region of the brain.
Loudspeaker parameter determination
Authors: Carlson David E.
Published: October 2005

The determination of the electrical, mechanical, and acoustical properties of a given loudspeaker is a necessity for the audio engineer whose job it is to design an enclosure for that loudspeaker. In the past, methods of measurement for determining the various characteristic parameters of a loudspeaker have often been quite complicated and time consuming, and sometimes subject to experimented error. A new measurement and calculation technique has been developed with an emphasis on the minimization of these experimental errors and the time consumption necessary for the procedure. This new method relies heavily on past loudspeaker analysis and theory, and represents somewhat of a hybrid model of current state of the art loudspeaker parameter-determination techniques.
Concepts in nonlinear acoustics: I. Fundamental concepts
Authors: Pederson Thomas B.
Published: October 2005

In the course of deriving the linear wave equation, one is required to “linearize” the constitutive equations by assuming that during the passage of the disturbance the various parameters involved undergo infinitesimally small perturbations from their static values. But what of the case in which the disturbance is large enough such that the perturbations cannot be safely considered infinitesimal? Obviously in such a case the linear wave equation no longer holds and another equation, nonlinear in nature, will result. In the past decade, this equation has taken on increasing importance to a relatively new field called “nonlinear acoustics.” This paper presents a few of the concepts of the field, stressing mainly the even more specialized field of parametric sonar, which makes use of the peculiarities involved in nonlinear acoustics.
Canadian activities in community noise measurements
Authors: Benwell D. A.
Published: October 2005

As part of the federal government's role to protect Canadians from the adverse effects of noise, an assessment of noise hazard and control has recently been completed and will be published. It is planned to use this document, together with further studies to establish criteria on the auditory and nonauditory effects of noise. Community noise studies in Canada have been assessed as part of this work, particularly for their usefulness toward government regulations and other noise control activities. Descriptions of community noise measurements in Alberta, British Columbia, and Ontario will be amongst those discussed. Emphasis will be given to instrumentation, techniques of analysis and reporting, and temporal and spatial sampling used.
Evaluation of temporal sampling schemes
Authors: Corley D. M. , Mansbach P. A. , Pallett D. S.
Published: October 2005

Four community sites were monitored continuously for periods of from one to three weeks. A-weighted sound pressure levels were digitally recorded every 0.1 sec, wind data were recorded simultaneously. Proposed temporal sampling schemes may be efficiently “tested” by processing this data on a high-speed digital computer, comparing the results obtained by using the sampling schemes with results computed from the entire data set. A priori predictions of accuracy of the sampling schemes are compared with the accuracy actually achieved in this experiment. Results using several familiar sampling techniques are presented.
Physical attributes of traffic noise which correlate with human perception and preference
Authors: Cermak G. W.
Published: October 2005

Human subjects made choice and dissimilarity judgments about pairs of 1-min recorded traffic sounds which had been equated for their energy-equivalent sound level (Leq). Leq had been found, in a previous experiment, to be the dominant noise measure determining such judgment data when the same traffic sounds were not equated in Leq [G.W. Cermak and P.C. Cornillon, J. Acoust. Soc. Am. 59, 1412–1420 (1976)]. The judgment data were significantly different than those from the previous experiment, indicating that Leq is not the only variable which can be used as a basis for choice and dissimilarity judgments of traffic sounds. The physical measure of the set of traffic sounds which correlated best with the judgment data was the time-dependent variability of dBA (the mean absolute difference between adjacent terms in a time series of 0.34-sec dBA values). The standard deviation of the dBA levels did not correlate significantly with the judgment data.
Review of acoustic lens developments
Authors: Folds D. L.
Published: October 2005

Use of acoustic lenses in underwater applications spans many decades. Lord Rayleigh describes an acoustic lens as early as 1879. Some well-documented experimental lenses were developed by Williams in the 1940's at General Electric. This work with lenses and reflectors led to the development of the U. S. Navy's first high-frequency (100-kHz) sonar, the AN/UQS-1. Since that time many investigators have worked in the area developing liquid, solid, and gas lenses. The complexity of these lenses range from the simple single-element lens to a recently developed three-group seven-element lens. These lens developments will be reviewed and the current state of lens technology will be discussed.
A Tee-network, containing negative components, suitable for simulating distributed mass-stiffness systems
Authors: Miller Harry B.
Published: October 2005

An obstacle existed in the past to simulating distributed parameters by means of a Tee-network. This obstacle was the requirement to represent impedance functions having negative slope, such as — cosecant wl/c. Such a representation is not possible with passive elements. However, this can now be easily done in several ways. One way is to use lumped impedance elements combined with active circuits, to produce negative components. Another way is to use a desk-top calculator in conjunction with an equivalent-circuit analysis program that uses lumped elements, positive or negative. People who like circuits obtain a certain additional insight from this circuit approach, an insight which is not given by the usual analysis methods. The method is applicable to electromechanical transducer simulation, vocal-tract simulation, and woodwind instrument simulation. Some graphical solutions to a transducer problem, automatically obtained in a few minutes, will be shown.
Auditory handicap of hearing impairment and the limited benefit of hearing aids
Authors: Plomp R.
Published: October 2005

The aim of this paper is to promote a better understanding of hearing impairment as a communicative handicap, primarily in noisy enviornments, and to explain the essentially limited applicability of the hearing aid. From available data we can conclude that about 7.5% of the population is auditory handicapped, with a hearing loss for speech ⩾ 24 dB. Every hearing loss can be interpreted as the sum of a loss class A (Attenuation), characterized by a reduction of the levels of both speech signal and noise, and a loss class D (Distortion), comparable with a decrease in speech-to-noise ratio. On the average, the hearing loss of class D (= hearing loss in noise) appears to be about one-third (in dB) of the total hearing loss (A + D = hearing loss in quiet). A hearing aid can compensate class-A hearing losses, giving difficulties primarily in quiet, but not class-D hearing losses, giving difficulties primarily in noise. The latter class represents the first stage of auditory handicap, beginning at an average hearing loss of about 24 dB. An algorithm for computing speech level relative to speech-reception threshold as a function of hearing losses of classes A and D, as well as acoustic gain, internal noise, and quality of the hearing aid, has been developed and examples will be given.
Comparison of a two-branch and a four-branch artificial ear
Authors: Studebaker G. A. , Cox R. M.
Published: October 2005

A two-branch artificial ear was compared with two four-branch artificial ears (Zwislocki couplers). Overall results from the two coupler types were very similar. The equivalent volumes of the couplers were evaluated by comparing the levels developed in these couplers with the level developed by the same signal in a 2-cc cavity. From 100 to 5000 Hz the equivalent volume of the two-branch coupler was slightly less than that of the four-branch couplers, the difference never exceeding 0.1 cc. The acoustic damping provided by the couplers was compared by measuring the amplitude of reactance resonances observed when the couplers constituted the volume of the resonator (a situation analogous to the acoustic system operating when vented earmolds are used). In the 400–700-Hz range, resonances developed in the two-branch coupler were 2–3 dB less damped than the analogous four-branch coupler resonances. In the 2–3-kHz range the damping of the couplers was approximately equal. This evidence suggests that the two-branch coupler provides less damping of typical vent related resonances than does the real ear. [Work supported by NIH.]
Harmonic configuration—the effect on the “hearing out” process
Authors: Gilliom J. D. , Cline C. , Taylor D. W.
Published: October 2005

Is the consonant or dissonant characteristics of a musical chord related to the degree to which individual components of a chord can be separately “heard out”? To approach this question, a paradigm was used which required the observer to extract frequency information from the middle component of a three-component chord in order to perform efficiently in a simple monaural yes—no detection task. The observers's task was to detect a 100-msec sinusoid in continuous Gaussian noise with the signal frequency being one of four possible frequencies. To reduce this frequency uncertainty, a 300-msec contralateral cue beginning 700 msec before the potential signal event was available on each trial. The cue consisted of three components with the middle component always matching the frequency of the potential signal event. The harmonic configuration of the overall cue was varied randomly from trial to trial across four harmonic relationships. The results indicate that the harmonic configuration of a chord significantly influences the observer's ability to extract information from components of that chord.
Analysis and effect of design in differential sensitivity to musical frequencies and intensities in auditoria
Authors: Mulder George
Published: October 2005

A study of the application of music-acoustical principles including absorptive materials, reactive elements, baffles, mufflers, sound silencers, etc., in the Corson Auditorium of the Grand Traverse Performing Arts Center, The Interlochen Center for the Arts, Michigan, comparing especially the intensity and frequency, the pitch and timbre of pianos and other keyboard stringed instruments. The evoked responses to anechoic chamber design and the physiological as well as psychological attitudes of the subjects were utilized in these studies. Some measure of reassessment of physical features to psychological responses of musical understanding may be forthcoming. However, the results of these direct acoustical estimations and provisions agreed almost perfectly with earlier assumptions.
Loudness may adapt: after all
Authors: Fishken D. , Arpino E. , Testa M. , Scharf B.
Published: October 2005

A number of recent studies have demonstrated that, in the absence of contralateral stimulation, the loudness of a continuous sound does not diminish over time. Using the methods of magnitude estimation (ME) and constant stimuli (CS), we also find no loudness decrease (i.e., no adaptation) at sensation levels between about 30 and 70 dB. However, below 30 dB SL, the loudness of a tone at 1 or kHz presented either monaurally or binaurally, diminishes dramatically, the more, so, the lower the level. In ME, observers assigned a number to the loudness of a steady tone at 18, 36, 72, and 100 sec after onset. In CS, observers judged whether a 1-sec tone in the left ear was louder or softer than a tone in the right ear that ended 400 msec earlier and that had lasted either 1 or 30 sec. An earlier study by Margolis and Wiley [J. Acoust. Soc. Am. 59, 222–224 (1976)] showed no adaptation by the method of CS at 25 dB SL whereas we measured average adaptation of 13 dB. Reasons for this descrepancy remain unclear, but we and the other two dozen observers in our experiments do agree among ourselves that soft tones become still softer over time intervals up to at least 100 sec.
Monaural and binaural temporal gap detection
Authors: Pollack Irwin
Published: October 2005

Temporal gap thresholds were obtained for four combinations: presentation of monopolar (M) versus random-polarity (R) pulse trains; and, presentation of signals before and after a temporal gap to the same (S) ear or to different (D) ears. Only when a spectral analysis of the signals and gaps is possible (condition M-S is extremely fine temporal gap sensitivity obtained (∼1 μsec). As expected with spectral processing, temporal gap sensitivity improves with longer and longer signals on each side of the gap. Sensitivity decreases markedly for the other conditions (about 103-fold for condition R-S; and, about 104-fold for condition M-D or R-D). Under these conditions, gap sensitivity decreases with longer and longer signals around the temporal gap. The poorer sensitivity with longer signals for binaural gap detection may be another representation of the binaural “precedence” effect. Strong asymmetrical effects upon gap detection were noted.
Should an informational cue precede or follow the signal event: results with a contralateral cue in an uncertain frequency paradigm
Authors: Gilliom J. D. , Taylor D. W. , Cline C.
Published: October 2005

This study continues the investigation of factors influencing the effective use of an informational cue. The observer's task was to perform a monaural yes—no detection task with a 0.5 probability of a signal event (a 100-msec sinusoid in continuous Gaussian noise whose frequency was chosen randomly from an ensemble of eight frequencies). On each trial, a 300-msec contralateral cue matching the signal frequency was provided. The observer was encouraged to use this cue to reduce the effects of the existent signal frequency uncertainty. When the starting time of the cue relative to the signal event was varied across experimental conditions: (1) Given a sufficiently long interstimulus interval, the cue could be effectively utilized whether placed before or after the signal event. (2) A cue presented before the signal event produced a larger improvement in performance than a cue presented after the signal event. (3) A cue presented with a small interstimulus interval produced substantial decrements in detection performance.
Some developmental aspects of the acoustic reflex
Authors: Hopkinson N. T.
Published: October 2005

Acoustic-reflex measurements were conducted on more than 500 subjects age three to six years old. Presence or absence of the reflex at 110 dB HI was the criterion measure at frequencies of 1000, 2000, 4000 Hz and a low-pass, high-pass, and broadband noise. The probe tone was 220 Hz. If the reflex was absent at any one or all of the above stimuli in either or both ears, the subject was categorized for absent reflexes. Absent reflexes were analyzed with regard to variables such as age, sex, race and socioeconomic level over one season. General findings were as follows: The acoustic reflex to broadband noise was elicited for every subject in the group but was absent for either high- or low-pass noise or both in 2.8% of the subjects. Acoustic reflexes were absent at 4000 Hz alone in 9.2% of the subjects. Only 2.3% of this group of subjects could not be tested for the acoustic reflex. [Work supported by NINCDS.]
Infant place discrimination of voiced stop consonants with and without release bursts
Authors: Bush Marcia A. , Williams Katherine L.
Published: October 2005

A number of studies have shown that infants as young as one month of age are able to detect changes in place of articulation among the voiced stop consonants in English [Moffitt, 1971; Morse, 1972; Eimas, 1974]. In each of these studies, the stimuli were three-formant synthetic patterns in which the desired place changes were realized by manipulating the starting frequencies of the F2 and F3 transitions. The present experiment was designed to examine the infant's ability to use the additional acoustic information provided by the consonantal release burst in making the /d-g/ place distinction. Infants between six and twelve weeks of age were tested using the HAS (high-amplitude sucking) procedure. Two pairs of synthetic, five-formant /da/-/ga/ syllables (with and without the appropriate d or g bursts) were used as stimuli. The experimental data show that, although infants are able to discriminate both sets of syllables, evidence for discrimination is greater for stimuli with bursts. This finding suggests that increasing the acoustic contrast between two stimuli will facilitate place discrimination by infants—perhaps by enhancing the integrated spectral properties characteristic of the individual stop consonants. [Work supported by NICHD.]
Cross-language study of speech pattern learning
Authors: Simon C.
Published: October 2005

In order to investigate the nature of some processes in speech acquisition, synthetic speechlike stimuli were played to groups of English and French children between 2 and 14 years of age. The variable acoustic parameters were voice-onset time and first-formant transition. Three stages were observed in the development of children's labelling behavior. These were called scattered labeling, progressive labeling, and categorical labeling, repectively. Individual response patterns were examined. The first stage (scattered labeling) includes mostly children up to 2 or 3 years of age for the English, and up to about 4–5 for the French. Progressive labelling behavior is found mostly amongst children aged 3 and 4 (for the English), up to about 7–8 for the French. Categorical labeling only becomes the dominant pattern at the age of 5–6 for the English, several years later for the French. Not all follow this process with identical chronological stages. For instance, some 3-year-olds give categorical labeling and about one third of 11- to 13-year-olds do not respond in a sharply categorical fashion. However, the overall development was found to be highly significant (p smaller than 0.002, one tail test, for both English and French, using Kendall's Tau measure of correlation). These findings indicate that categorical labeling for speech sounds is not innate but learned through a relatively slow process which is to a certain extent language specific. The interpretation of previous experimental work in the field may have to be re-examined—as well as their theoretical implications—in the light of the present results.
Binaural interaction in consonant perception
Authors: Yonovitz A. , Lozar J. T. , Thompson C. L. , Bickford Barbara J.
Published: October 2005

It is generally known that subjects' binaurally masked threshold improves when either the signal or noise is presented to the two ears antiphasically. The discriminability of 21 initial consonants (CV's with 'a) were studed under three conditions: both the stimulus and masker in phase (SoNo), stimulus in phase and masker antiphasic (SπNo), and the stimulus antiphasic with the masker in phase (SπNo). At various signal to noise ratios subjects wrote their response to 30 random presentations of each CV under each phasic condition. From 630 responses per subject confusion matrices were constructed. Data were analyzed in terms of consonant intelligibility, consonant confusions, and distinctive feature intelligibility. Information transmission values indicated differences in the contribution of the various features with different phasic conditions and also revealed changes in this pattern as a function of signal-to-noise ratio.
Cross language differences in tone perception
Authors: Gandour J. , Harshman R.
Published: October 2005

Using an individual differences multidimensional scaling model of perception, the present crosslanguage investigation seeks to determine what dimensions underlie the perception of linguistic tone, and to what extent an individual's language background (Thai, Yoruba, or American English) influences his perception. Dissimilarities data were obtained from subject's paired—comparison judgments of thirteen different pitch patterns superimposed on a synthetic speechlike syllable. A multidimensional scaling analysis of the data for the total group revealed that five dimensions—interpretively labeled, AVERAGE PITCH, DIRECTION, LENGTH, EXTREME ENDPOINT and SLOPE—best summarize the perceptual structure underlying the dissimilarities data. Language subgroup variation in relative importance of these dimensions appears to be primarily related to subgroup differences in the way pitch is used to convey linguistic information. Discriminant analysis showed that most individual speakers of a tone language (Thai or Yoruba) can be easily distinguished from speakers of a nontone language (English) on the basis of their distinctive patterns of perceptual saliency for these five dimensions. [Research supported in part by fellowship grants provided by SSRC and NSF to first author. UCLA 1975–1976.]
Identification of sine-wave analogues of CV syllables in speech and nonspeech modes
Authors: Bailey Peter , Dorman Michael , Summerfield Quentin
Published: October 2005

Adults discriminate categorically successive from simultaneous coterminous acoustic events—a psychoacoustic capability which may account for infants' sensitivity to contrasts in stop-consonant voicing. Can adults demonstrate an analogous capability to explain infants' sensitivity to contrasts in stop place? We synthesized a set of [vV-dV] “asymmetric” speech continua whose phoneme boundaries did not correspond to any obvious discontinuity:boundaries were not characterized by level formant transition trajectories nor did they fall at the centers of stimulus ranges. Each syllable was modelled by replacing the first three formants with frequency and amplitude modulated sine waves. A modified AXP discrimination paradigm in which X was a member of a continuum and A and B were continuum endpoints provided “identification” data for both types of stimulus. Initially, sine-wave continua were categorized symmetrically while their constituents were perceived as “whistles.” In later sessions, however, listeners volunteered that these patterns afforded consonantal percepts and categorized both sine-wave and formant continua asymmetrically. We are continuing to experiment with this paradigm which provides identification data without requiring overt labelling. We note that our present results provide no evidence of a psychoacoustic basis for sensitivity to place of production contrasts.
Synthesizing whistle stops
Authors: Bernstein Jared
Published: October 2005

Certain human languages have a derived form in which sound is generated by whistling. The information found in syllabic portions of speech is usually conveyed in whistle language by the frequency of the whistle as it mimics either the fundamental frequency (for tones) or the second formant (for vowel qualities) in speech. Some of these whistle languages include consonant articulations. This paper describes techniques for hand synthesis of whistled [t] and [k] and automatic analysis—synthesis of whistle stops using Wm. Henke's MITSYN system. Characteristic bursts and frequency transitions for stops during bilabial whistling are discussed and compared to bursts and transitions in speech. [Work supported by NIH Grant NS 04332.]
Packet transmission of speech using variable-quality coding and time-interval modification
Authors: Webber S. A. , Harris C. J. , Flanagan J. L.
Published: October 2005

Speech transmission by switched digital packets provides opportunities for increased utilization of transmission capacity. We describe a combination of processing features designed to maximally load a transmission facility and accommodate fluctuating demands on it. Our system detects individual speech energy bursts and, in effect, discards the silent intervals between bursts. It digitally encodes each burst with a quality (bit rate) that reflects serve demands being made on the transmission facility at the moment. In a preliminary study of these features, we find that elimination of within-sentence silent intervals saves about 20% of the total sentence time. We also find that digital coding (by ADPCM) in the range 40–20 Kbits/sec provides useful quality variation. We suggest that these flexibilities and modest buffer storage can aid utilization of transmission capacity.
Analysis of the performance of a tandem connection of a 2.4-kB/sec LPC vocoder to a 16-kB/sec CVSD coder
Authors: Crochiere R. E. , Goodman D. J. , Rabiner L. R. , Sambur M. R.
Published: October 2005

The performance of a tandem connection of narrow-band and wide-band speech communication systems is evaluated. Specifically, the narrow-band system consists of a conventional LPC vocoder operating at a bit rate of 2.4 Kb/sec and the wideband system consists of a CVSD (continuously variable slope delta modulator) operating at a bit rate of 16 Kb/sec. This paper describes the properties of the narrowband to wideband link. The SNR (signal-to-quantizing noise ratio) of the CVSD coder is analyzed over a 50-dB variation of the input signal levels and for a variety of source excitations for the LPC synthesizer. It is shown that SNR improvements in the CVSD coder of 2–2.5 dB are possible in the slope overload region of the coder by modifying the source excitation of the LPC synthesizer and by preprocessing the input signal to the coder with an allpass filter. Both methods aid in reducing the peak factor (peak to rms level) of the input speech to the coder. Subjectively, however, only slight improvements in quality, if any, were observed with these modifications. The weakest link in terms of quality in the tandem connection appears to be that of the CVSD coder.
Principal-components analysis of speech spectra
Authors: Zahorian Stephen , Rothenberg Martin
Published: October 2005

The principal-components method is sometimes employed to describe speech spectra in terms of a small number of factors, that is eigenvectors of the covariance matrix. In this study, this method has been used to analyze the spectral properties of continuous speech and to regenerate the spectra from a small set of factors. Spectral data from 0 to 5000 Hz was obtained using a linear-predictive (LP) all-pole model. This data was used to calculate the average energy in twenty bands, each about an auditory critical bandwidth wide. The analysis was applied to (1) the log-coded band energies and (2) the band energies to the one-third power. Continuous speech was synthesized using spectra computed from a small number of principal-component factors and the LP residual signal. The quality of this speech is compared with speech synthesized using a low-order LP vocoder. In addition, it is argued that the principal components might be more easily identifiable with linguistic categories than are the poles of the low-order LP model.
Underwater speech intelligibility as a function of fundamental frequency, speech rate, and intensity
Authors: Hicks J. W.
Published: October 2005

Previous investigations have demonstrated that divers can improve their speech intelligibility in the underwater milieu. However, little or no information specifying the parameters involved in such improvement is available. This study attempts to objectively evaluate the accuracy of divers' ability to systematically and independently manipulate the speech parameters of intensity, rate, and speaking fundamental frequency and to study the effect of these changes on speech intelligibility. Divers were trained to vary each speech characteristic individually (high and low levels for intensity and speaking fundamental frequency; fast and slow rates) while reading Griffiths word lists. Voice recordings were made at depths of 0, 30, and 300 ft. Intelligibility scores were obtained via standardized listening sessions carried out at the Institute for Advanced Study of the Communication Processes. The results indicate that the diver/subjects were able to manipulate the speech parameters with a good degree of accuracy, i. e., shifting each parameter in the correct direction while holding the others constant. While there was a tendency for low f0, high intensity, and slow rate to improve intelligibility, the magnitudes of these trends were not statistically significant.