bluuebunny/crossref_metadata_2025 huggingface dataset:
Dental flaps and vowel duration in English
Authors: Fox Robert A. , Terbeek Dale
Published: October 2005

This experimental investigated dental flaps and their effect upon preceding vowel duration in English. Twenty speakers read a randomized list of 40 disyllabic words, 20 with medial /d/. Most tokens of /t/ and /d/ were pronounced as flaps, as expected. The data, collected from spectrograms, included 1st and 2nd vowel duration, presence or absence of flap (as opposed to stop [t] or [d], duration of consonant, and extent of voicing during the flap. Our results provide evidence bearing on two phonological questions: (1) Does the rule which changes /t/ and /d/ to flaps (F) precede or follow the rule which lengthens vowels before voiced consonants (VL)? If VL precedes F, the vowels preceding flaps with underlying /d/ will have greater duration than flaps with underlying /t/. If F precedes VL, there will be no such difference. (2) Do underlying /t/ and /d/, in fact, give rise to the same voiced flap consonant, as is usually assumed? The data suggests that not all /t/ flaps and /d/ flaps are identical, nor are they necessarily voiced.
Linear prediction analysis of speech based on pole-zero representation
Authors: Atal B. S. , Schoeder M. R.
Published: October 2005

Speech analysis and synthesis by linear prediction is based on the assumption that the short-time spectral envelope of speech can be represented by a number of poles. An all-pole representation does not provide an accurate description of speech spectra, particularly for nasal and nasalized sounds. In this paper, a method is presented for characterizing speech in terms of the parameters of a pole-zero model. The basic approach used in the method is similar to one used for the all-pole case: the parameters of the pole-zero model are obtained by minimizing the mean-squared prediction error over the analysis interval. It is shown that, although the equations resulting from the minimization are nonlinear, both poles and zeros can be determined without using an iteractive procedure. The method has been applied to real speech data and the results show that the speech spectra derived from the pole-zero model agree very closely with the actual spectra derived by direct FFT analysis. Audible distortion due to inadequate treatment of zeros in the all-pole case disappear with pole-zero representation.
Duration of English stops in prestressed position
Authors: Zue Victor W.
Published: October 2005

Durational characteristics of English stops in prestressed position, both in singleton and in clusters, are examined in detail using a data base designed for controlled acoustic phonetic research. [Victor W. Zue, “Acoustic phonetic data base for the study of selected English Constants, consonant clusters, and vowels,” J. Acoust. Soc. Am. 57, S34(A) (1975)]. The study being presented here is based on the analysis of some 1200 stops spoke by 3 male speakers. Results indicate systematic variations of closure, voice-onset time, frication, and aspiration durations as a function of: (1) the voicing characteristics of the stop; (2) the place of articulation of the stop; and (3) the phonetic environment (i.e., the influence of the following vowel for singleton stops, and the influence of the preceding fricative and/or the following sonorant for clusters). Results on inter- and intraspeaker variability will also be discussed.
Speech analysis by homomorphonic prediction
Authors: Kopec G. E. , Oppenheim A. V. , Tribolet J. M.
Published: October 2005

Linear prediction (LPC) is a generally accepted method for obtaining all-pole speech representations. However, in many situations (e.g., nasalization studies) spectral zeros are important and a more general modeling procedure is required. Unfortunately the need for pitch synchronization has limited the success of available techniques. This talk deals with a new approach to pole-zero identification, based on homomorphic prediction, which avoids the synchronization problem. Homomorphonic prediction is a three step process. Homomorphic deconvolution is used to obtain a minimum-phase estimate of the vocal tract impulse response. Such a signal by definition is properly time synchronized. Linear prediction is applied to this waveform to identify its poles. Next, the LPC “residual” (error signal) is computed by inverse filtering. This signal contains the information about the zeros. Its z transform is approximated by a polynomial either through a weighted least-squares procedure (as in Shanks' method) or by spectral inversion followed by a second evaluation on real speech data will be presented. [Supported in part by ARPA, NSF, and the Fannie and John Hertz Fonudation.]
On detecting nasals in continuous speech
Authors: Mermelstein Paul
Published: October 2005

The acoustic manifestation of nasal murmurs is significantly context dependent. To what extent can the class of nasals be automatically context? This contribution reports on the characterization of the spectral change accompanying the transition between vowel and nasal for the purpose of automatic detection of nasal murmurs. The speech is first segmented into syllablesized units, the voiced sonorant region within the syllable is delimited, and the points of maximal spectral change on either side of syllabic peak are hypothesized to be potential nasal transitions. Four simply extractible acoustic parameters, the relative energy change in the frequency bands 0–1, 1–2, and 2–5 kHz, and the frequency centroid of the 0–500-Hz band at four points in time spaced 12.8 msec apart are used to represent the dynamic transition. Categorization of the transitions using multivariate statistics on some 524 transition segments from data of two speakers resulted in a 91% correct nasal/nonnasal decision rate. [Work supported by ARPA, Department of Defense.]
COSH measure for speech processing
Authors: Gray A. H. , Markel J. D.
Published: October 2005

Distance measures have been applied in such tasks as speaker verification, recognition, and variable frame rate transmission in vocoder systems. The DELCO algorithm, also known as the prediction residual or log likelihood ratio, has recently been applied in the abovementioned tasks. Although this procedure has been demonstrated to be of values in speech processing, (1) it does not satisfy from symmetry requirement for distance, i.e., the diatance from x to y is not equal to the distance from y to x, and (2) relationships between this measure and possibly more basic and physically meaningful measures, such as the root-mean-square (rms) difference between two log spectra, have not been shown. In this paper, the relationship between the likelihood measure and the rms measure is discussed and the nonsymmetrical property is illustrated. A cosh measure, related to two nonsymmetrical likelihood ratios, is then introduced with the following properties: (1) the symmetrical requirement for distance is satisfied, (2) its value, lying between the rms and peak value, can be related to log spectral differences, with large differences being weighted more significantly, (3) it equals the rms measure for small differences, and (4) it can be recursively evaluated using linear prediction techniques without FFT calculation. The nonuniform weighting property is believed to be of importance in speech processing since larger spectral differences generally occur near the formant frequencies.
Cepstral distance and the frequency domain
Authors: Markel J. D. , Gray A. H.
Published: October 2005

Atal [J Acoust. Soc. Am. 55, 1304 (1974)] recently investigated weighted Euclidean distance measures for speaker identification and verification tasks. The cepstral coefficients of the filter A(z) determined through linear prediction analysis resulted in higher scores than other parameters such as predictor coefficients or area functions. The purpose of this paper is to discuss several properties of the cepstral coefficients obtained from A(z) and to suggest that the results illustrate the importance of the smoothed log spectrum as a basis for distance measurements in speech processing. From statistical experiments on speech, it is shown that for very small cepstral lengths (on the order of M, where M is the number of filter coefficients), there exists a very high correlation between the ceptral measure and the root-mean-square (rms) Euclidean distance measure between two log spectral models obtained using linear prediction analysis. Theoretically, an infinite number of cepstral coefficients are necessary for an equality. Since the cepstrum can be recursively evaluated from the filter coefficients [if 1/A(z) is stable], the cepstral distance measure can be viewed as an efficient and accurate replacement for the direct root-mean-square (rms) measure between two log spectra but without FFT or logarithm operations.
Quantitative acoustic characteristics of fricative and stop consonant allophones
Authors: MacIver Baker Janet
Published: October 2005

It is well known that the acoustic manifestation of any phone is, to varying degrees, a function of its acoustic environment. We have tried to evaluate this explicitly. Our data consisted of all common fricative and stop consonant allophones embedded in carrier phrases, each rendered by three speakers (two male, one female). Time-domain segmentation and analysis, based on individual homogenous segments of frication and release-aspiration, from the fricatives and stops, respectively, and characterized these by six parameters. For each phoneme and speaker, each allophone was normalized with respect to an appropriate neutral referent allophone. Pairwise phoneme recognition tests tried to distinguish to which of 2 phonemes belonged an unknown allophone of one of these. A modified Euclidean distance measure compared the acoustic characteristics of this allophone with the predicted allophone effects derived from all allophone, belonging to /k/ or /p/, was assigned to one of these on the basis of allophone characteristics derived from /g/, /b/, /d/, and /t/, by the same speaker. The ranges of scores for the 3 speakers, for different and same places of articulation, were 71%–86% and 44%–69% for stops, and 83%–95% and 75%–83%, for fricatives, respectively. [Part of work supported by ARPA (♯F44620-73-C-0074), at Carnegie-Mellon University.]
Use of scale model to design vehicle pass-by noise test facility enclosure
Authors: Gerhold C. H. , Tree D. R.
Published: October 2005

Inclement weather can cause up to 50% downtime for a vehicle pass-by noise measurement facility. One way to increase the facility efficiency is to shelter the test pad by an enclosure. Such an enclosure must be designed acoustically to minimize the effect that the enclosure structure has on altering the hemispherical sound field of the pass-by facility. This acoustic design is developed accurately and relatively inexpensively by tests on a scale model of the enclosure. The model structure is a replica of the proposed enclosure. The vehicle is modelled by a point noise source fixed at positions to simulate the source-receiver orientations of the major noise sources of a vehicle during drive-by. Each acoustic design is evaluated by comparison of sound pressure level spectra measured within the structure to spectra gathered in the semifree acoustic field. The spectra of the final acoustic design generally agree with the semifree field spectra to within ±1 − 12 dB. A properly designed enclosure could double the usable time of a vehicle pass-by noise measurement facility. By tests on a scale model, a feasible design has been developed.
Overhead microphone facility for recording vertically radiated vehicle noise
Authors: Oswald L. J. , Hickling R.
Published: October 2005

A description is given of an overhead microphone facility recently constructed at the GM Proving Ground in Milford, Michigan. It consists of an array of microphones suspended in a vertical plane from a cable that is stretched between two free-standing radio-communication towers. The cable spans a vehicle-noise passby-pad and the positions of the microphones in the array can be adjusted both vertically and horizontally. Important design features and considerations are discussed. Some sample results are presented for a city transit coach and two passenger cars during various modes of operation showing noise radiation patterns in vertical planes. The facility will be used not only in research areas relating to the study of motor-vehicle noise sources, but also to research areas related to outdoor sound propagation and community noise.
Detectability and effectiveness of automotive warning signals
Authors: Fidell S.
Published: October 2005

A method is described for applying the psychophysical Theory of Signal Detectability to predictions of ranges at which audible warning signals from emergency vehicles are detectable by drivers of cars. Isodetectability contours are defined for both the signal source and driver; sample contours are presented, and their utility in quantifying the effectiveness of warning signals is discussed. Comment is also made on the contribution to community noise exposure of audible warning signals and on means for minimizing annoyance.
Optimal shading of the general line array
Authors: Streit R. L.
Published: October 2005

Optimal synthesis for the general line array is defined here to mean the determination of element currents to minimize the sidelobe level for a specified beamwidth. Presently, available methods for the solution of this synthesis problem rely on either (1) a special set of positions (equispaced or symmetrically spaced) for which globally optimum currents can be computed or (2) some mathematical programming method (e.g., gradient descent) which can give only locally optimum results. In this paper, the author gives an algorithm which will yield globally optimum element currents for any given set of element positions. The first step of this algorithm maximizes the ratio of the power in the mainlobe region to the power in the sidelobe region and may be computed explicitly, since it can be expressed as an eigenproblem. Each additional step in the algorithm requires an iterative solution which may not be worth the effort, since the first step alone yields excellent results. The algorithm has the advantage of being applicable to broadband synthesis and the consequent suppression of grating lobes whenever the specified element positions make that possible. The primary drawback is that some sets of element positions result in numerically ill-conditioned eigenproblems. An example is given to illustrate the algorithm. [This work was sponsored by the Naval Underwater Systems Center.]
Resolution capabilities of passive sonar receiving arrays
Authors: Murphy D. A.
Published: October 2005

Resolution is defined as a binary hypothesis test or detection problem using the formulation of Rockmore and Bershad [J. Acoust. Soc. Am. 58, 684–692 (1975).] The formulation assumes Gaussian random signal and noise. The sufficient statistic, from the likelihood ratio, is compared with a threshold to determine whether the signal is from one or two point sources with the total power constant for the two hypotheses. The noise covariance is general, the sources may be in the nearfield of the receiving array, and the receiving array is general. The resolution problem is a special case of the discrimination problem with interference, covered by Rockmore and Bershad. A simple receiving array consisting of only two point receivers with broadside geometry both for sources and receivers is used to simplify the general expression and show its functional dependence on wavelength, source and receiver apertures, range, and signal-to-noise ratio. The resulting expression for the output signal-to-noise ratio, which measures the performance, is identical to that of Rayleigh multiplied by a constant and the signal-to-noise ratio at the receiver. [Work supported by ONR.]
Effects of intravehicular noise on electromyographic activity associated with the braking reflex
Authors: Bhéreur P. , Mergler-Racine D.
Published: October 2005

Previous results showed braking reaction times (Rt's) increased with higher intensity intravehicular noise (IN). Experiments were performed to relate electromyographic (EMG) activity to these RT changes. IN, recorded in a moving car, was reproduced at 77 and 85 dB, through a calibrated sound system, to subjects in a simulated vehicle. During 3-h exposures, subjects were required to maintain a fixed speed on the speedometer and brake when a red light lit up at random intervals. Surface electrodes recorded EMG activity from right gastrocnemius and rectus femoris muscles. Digital counters provided the braking reflex Rt's. For the rectus, at 77 dB, following a latency of 0.1–O.3 sec, there was a burst of activity which tapered off. With time, the amplitude of the response diminished and its duration increased. At 85 dB, this latter pattern, generally associated with fatigue, appeared earlier in time than at 77 dB, with significantly longer duration. For the gastrocnemius triphasic response, there was a progressive decrease in activity of the 1st and 2nd components which occurred earlier at 85 rather than 77 dB. These findings suggest that higher intensity IN induces increasing motor fatigue.
Modeling the vibration response of the hand due to hand-induced vibration
Authors: Reynolds D. D. , Keith R. H.
Published: October 2005

The mechanical response characteristics of the hand due to vibration directed into the hand by a vibrating handle was modeled as a three-degree-of-freedom, mass-spring-damper system for vibration in each of three mutually orthogonal directions. The coupling that existed between the three mass elements and the relative values of the different components of the mechanical model correlated very well with the actual coupling and the physical relations that exist between the different layers of tissues in the fingers and hand. The model indicated that the mechanical response of the hand due to hand-induced vibration was a function of the direction of the vibration relative to the hand and a function of the manner in which the vibrating handle was clasped. The model implied that vibration directed into the hand at frequencies below 100 Hz was transmitted through the different layers of tissues in the hand into the skeletal system and then up the arm. Vibration energy at frequencies above 100 Hz tended to be isolated to the tissued directly in contact with the vibrating handle.
Recycled facilities for a music college
Authors: Cavanaugh W. J.
Published: October 2005

Acoustical design considerations are of paramount importance in the design of new music teaching, performance, and rehearsal facilities. When existing buildings previously used for other activities are recycled to house the diverse functions of a rapidly growing college for training contemporary musicians, the acoustical design problems are not only more complex but provide a unique challenge for the architect, the college administrators, and all the engineering disciplines involved. The architectural and engineering team of Kubitz and Pepi Inc. was faced with just such a challenge in the conversion of a former hotel and adjacent movie theater building to house the Berklee College in Boston. This paper reviews the room acoustics, sound isolation, mechanical system noise control, and “acoustical cross-talk” design considerations, as well as measurements in some of the completed portions of the building. The acoustical results are reviewed in terms of the ever present considerations of budget, aesthetics, and, in this instance, the problems of scheduling the construction so that maximum use of the facilities could be retained while the work proceeded and occupancy of completed spaces realized at the earliest possible dates.
Designing for visual cues that enhance speech reception
Authors: Boyer Lester L.
Published: October 2005

Design guidelines which incorporate the visual contribution to speech are developed for maximum room size and speaker-listener distance, and for minimum lighting level on the speaker, when unamplified speech level and background noise level are known. Room-scale studies on speech intelligibility are conducted at different speech-to-noise ratios and different levels of visual acuity. Visual acuity has not been considered as a factor in earlier studies on the visual contribution to speech [W. H. Sumby and I. Pollack, J. Acoust. Soc. Am. 26, 212–215 (1954)]. The articulation index is evaluated on the basis of the modified rhyme test in accordance with the newly proposed international standard on speech intelligibility testing. The studies confirm the striking effect of speech-to-noise ratio on intelligibility. Also, the contribution of visual cues is shown to be very considerable, especially when the level of audible speech at the receiver is near the background noise level. The amount of improvement in speech intelligibility is shown to depend upon the illumination level on the speaker and the distance from the speaker.
Acoustical considerations for structures with air-supported roofs
Authors: Coffeen Robert C. , Anderson Wendell D.
Published: October 2005

Air-supported roof systems for large facilities such as arenas and stadiums are relatively recent developments in the search for economical methods for enclosing such spaces. However, such roof systems provide acoustical problems relating to reverberation, focused reflections, exterior-to-interior noise reduction, and interior ambient noise levels as established by pressurizing blowers. This paper discussed these concerns with primary reference to the new Pontiac Metropolitan Stadium in Pontiac, MI, and the new Uni-Dome at the University of Northern Iowa, Cedar Falls, IO. For the Uni-Dome project, Ranger Farrell was the acoustical consultant.
Concert/recital hall acoustics: an objective/subjective review of the new Pick-Staiger Concert Hall at Northwestern University
Authors: Kirkegaard R. L.
Published: October 2005

Program uses ranging from solo recitals to full concert orchestras and marching bands and attendances ranging from sparse audiences to full-house capacities create grossly different acoustics requirements in a 1000-seat music school concert/recital hall. The new concert hall at Northwestern University represents a state-of-the-art solution to this acoustic challenge. Extensive objective studies and subjective results will be reviewed in relation to initial design criteria.
Architectural acoustics in America to 1930
Authors: Shankland Robert S.
Published: October 2005

The beginning of architectural acoustics as an engineering science began with Joseph Henry when he was requested by Congress to design a new lecture hall for the Smithsonian Institution in Washington. Henry's preliminary investigations led him to the discovery of what is now known as the “precedence effect,” setting limits on the permissible time lags in successive acoustic signals before echoing is evident. Henry's design of the lecture hall made skillful use of his discovery for the location and shape of wall and ceiling surfaces to produce much “early reflected sound” to enhance the direct impulses. The next great advance in architectural acoustics was made by Prof. Wallace C. Sabine of Harvard in his quantitative studies of reverberation, and in his masterful use of these discoveries in the acoustical designs for many notable buildings, especially Boston Symphony Hall—still considered one of the world's finest. After Sabine's untimely death in 1919 his work was extended and applied by many others in the next decade, notably, Dayton C. Miller, F. R. Watson, Paul E. Sabine, Carl Eyring, R. F. Norris, and Vern O. Knudsen. The early development and application of architectural acoustics as an exact science was primarily an achievement of the New World, and one of which we can be justly proud in the Bicentennial year.
Building acoustics in America, 1920–1940
Authors: Sabine Hale J.
Published: October 2005

The period following the pioneer work of W. C. Sabine was marked by widespread application of the basic principles of room acoustics and sound absorption to practical problems. Factors in this development were the building of laboratories for full scale acoustical testing of materials and structures, the extensive publication of textbooks, articles, and test data, the rapid growth of an acoustical materials industry, and the contributions to advancement and dissemination of the state of the art by acoustical consultants and laboratory directors. Important new developments during this period include the formulation of theory and practice for the measurement of sound transmission through walls and floors, the establishment of exact criteria for the intelligibility of speech in auditoriums, and the application of the concepts of acoustic impedance and normal modes to new theoretical models of the interaction of sound waves with room boundaries.
Preliminary investigation of speaker height and weight identification
Authors: Davis Margaret , Lass Norman J.
Published: October 2005

The purpose of this investigation was to determine if listeners were capable of speaker height and weight identification from recorded speech samples of speakers' voices. A standard prose passage was recorded by 30 speakers, 15 females and 15 males. A master tape containing the randomly arranged recorded readings of all speakers was played to a group of 30 subjects. All subjects participated in two experimental sessions. In one session they were asked to determine the height of each speaker and in another session weight judgments were made. The order of presentation of the two experimental tasks was randomized so that 15 subjects made height judgments first and 15 subjects made weight judgments first. A multiple-choice response sheet containing four choices for height and weight judgments for each speaker was provided. In addition, subjects were also asked to indicate the confidence of their decisions on a seven-point rating scale, and to describe the criteria upon which they based their decisions. Results indicate that (1) for both order groups, subjects were capable, with better than chance-guessing accuracy, of correctly identifying the heights and weights of speakers when presented with only recorded speech samples and (2) subjects exhibited slightly greater accuracy in identifying speaker weights than heights, regardless of the order in which they were exposed to the two kinds of identification tasks. Implications of these findings and suggestions for future research are discussed.
Effects of selected vocal disguises upon spectrographic speaker identification
Authors: Reich Alan R.
Published: October 2005

This research was designed to investigate the effects of selected vocal disguises upon spectrographic speaker identification. The experiment consisted of open-trial, spectrogram-matching tasks with 15 “reference” and 15 “matching” speakers. The speakers produced two sentence sets in six different ways: (1) normal speaking mode, (2) old-age disguise, (3) hoarse disguise, (4) hypernasal disguise, (5) slow-rate disguise, and (6) free disguise of his own choosing. The reference spectrograms were always undisguised speech samples; the matching spectrograms were either disguised or undisguised. Four trained spectrographic examiners completed the matching tasks. The results revealed that the examiners were able to match speakers with a moderate degree of accuracy (56.67% correct) when there was no attempt to vocally disguise either utterance. The inclusion of disguised speech samples in the matching tasks significantly interfered with speaker-identification performance. However, certain voice disguises were more effective than others. In addition, certain speakers were considerably more difficult to identify than others. The examiners exhibited poor test-retest reliability for the undisguised tasks. In summary, this investigation did not substantiate prior claims that spectrographic speaker identification is unaffected by attempts at disguising one's voice.
Observations of glottal waveforms in selected phonatory tasks
Authors: Colton R. H. , Estill J.
Published: October 2005

Glottal volume velocity waveforms during sustained phonation were estimated by inverse filtering the air flow waveforms recorded with a circumferentially vented face mask of a design described by Rothenberg [M. Rothenberg, J. Acoust. Soc. Am. 53, 1632–1645 (1973)]. Five adult males produced the vowel /a/ (1) in their typical every day speech, (2) as a cry or whine, (3) in a country-western twang, and (4) with an operatic ring. “Speech” waveforms exhibited slightly faster closing times with some dc flow during the closed phase. The waveforms of the “whine” are much simpler in shape with little or no closed time and large dc flows. During “twang” phonation, the waveforms exhibited longer closed times, sharp opening, and closing times with little dc flow. Evidence of nasalization was also noted. The waveforms of the operatic ring exhibited definite closed phases and fast closing time. Some waveforms illustrate the presence of the “singing formant” or concentration of energy at 3 kHz.
Air conduction + bone conduction = your own voice?
Authors: Saleeby Gary C. , Allen George D. , Mahaffey Robert B. , Wood Thomas J.
Published: October 2005

While subjects read a paragraph, a two-channel recording was made from an accelerometer strapped to the speaker's forehead (bone signal) and a microphone placed 5 cm from his ear (air signal). Each subject was then instructed to amplify or attenuate each channel on playback until the mixture most resembled his own voice. After adjustment, relative DBA measurements of bone vs air were taken from likewise prerecorded productions of isolated /n/ and /æ/. Intrasubject variability in these adjustments was low, indicating that the task could be performed. The mean-square intrasubject variability for /æ/ was, however, twice that for /n/, suggesting greater stability in the bone-air relationship of nasals. Intersubject variability was high and was three times greater for /æ/ than for /n/. The experimental system will be available for inspection and test at the session. Additional data will also be available regarding subjects' manipulations of the time relationship between the bone and air signals.
Syllable segmentation with time-domain variables
Authors: Todd Henry S.
Published: October 2005

The time-domain variables of cycle frequency and cycle peak-to-peak amplitude have been applied in a segmentation algorithm to determine acoustic syllable boundaries of continuous speech. The parameters are computed once for each cycle and are time averaged using a digital recursive filter. The long time average is used as a normalization parameter against which the short time average is compared. In addition, these parameters are compared against a decision threshold which varies with elapsed time from the last segmentation boundary. The results demonstrate that acoustic syllable units do not always coincide with linguistic syllable units. Where these two units are not in agreement, the acoustic syllable is usually longer. Additional segmentation of the acoustic syllable can likely be performed during syllable recognition.
Evaluation of a synthetic talker for speech intelligibility testing
Authors: Chial M. R.
Published: October 2005

Four experiments using initially naive, normal hearing subjects were conducted to determine limitations of the use of synthetic speech (Votrax VI) in clinical situations. All experiments used different random orderings of CID W-22 phonetically balanced word list 1 presented monaurally via earphones. Experiment I consisted of eight temporally contiguous trials with synthetic speech: learning effects across trials were minimal (7%), and different random orderings of stimuli produced virtually identical results (55% correct). Experiments II and III used subjects trained during Experiment I. In Experiment II, the sensation level of synthetic speech was varied from +5 to +30 dB, producing mean scores ranging from 29%–64%. In Experiment III, synthetic speech was presented at 70 dB (re 20 μPa) in the background of a competing wide-band noise. Signal-to-noise ratios varied from −5 to +20 dB. Mean scores ranged from 18%–57%, with asymptote at +15 dB. Experiment IV evaluated learning effects across eight trials in which human and synthetic renditions of test lists were alternated. Performance improved moderately for machine speech (10%–20%), but not for human talker materials (1%). Samples of synthetic speech will be presented and implications will be discussed.
Comparisons of hearing aid performance and speech intelligibility results in a standard and Zwislocki-type acoustic coupler
Authors: Smaldino J. J.
Published: October 2005

Electroacoustic measurements were made on ten hearing aids in both a standard 2-cc and Zwislocki-type acoustic coupler. Pearson correlation coefficients generally indicate that while measurements in the two couplers are different, the differences are systematic. Prediction equations relating electroacoustic measurements in the two couplers were generated. Several behavioral measures of speech intelligibility were recorded through the same aids in both couplers. Scores are presented and related to the systematic nature of the differences between the acoustic couplers.
Current problems in binaural hearing research
Authors: Colburn H. S. , Domnitz R. H. , Stern R. M. , Durlach N. I.
Published: October 2005

We will conduct an informal open discussion of issues related to stimulus control, choice and design of experiments, theoretical models, and general research strategy in binaural hearing. Topics to be considered include acoustical monitoring of stimuli, variability and drift in subject performance, techniques for determining the shape of binaural images, time-intensity trading, the relation of lateralization to detection, similarities and differences in current theoretical models, the inadequacy of many previous experiments to critically test and separate models, the incorporation of multiple sampling of inputs into models, and the effects of long-term previous experience in natural environments on the perception of laboratory stimuli (and the possibility of taking explicit account of such effects in the construction of models. [Work supported by NIH.]
Masking level differences: Auditory evoked responses with homophasic and antiphasic signal and noise
Authors: Thompson C. L. , Yonovitz A. , Lozar 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. It has similarly been shown that the amplitude of the AER varies as a function of the intensity and presumably therefore loudness of the stimulus. In this study, click stimuli were presented under three conditions: both the stimulus and the masker in phase (SoNo), stimulus in phase and masker antiphasic (soNπ), and the stimulus antiphasic with the masker in phase (SπNo). Subjects' AERs' were obtained at five intensity levels (re SoNo psychophysical threshold). Data are presented demonstrating the amplitude of the AER as a function of stimulus intensity for each of the three signal/noise phase conditions. While differences in the AER amplitudes are detectable for group data, individual responses show considerable variability.
Frequency dependence of lateral line microphonics
Authors: Boston J. R.
Published: October 2005

Because of the anatomical arrangement of the sensory hair cells, microphonic potentials recorded from the lateral line canal organs of fish in response to sinusoidal stimulation consist of a dc component and a second harmonic component. Several investigators have described these components and noted that they depend on frequency, but no systematic data have been presented. Experimental input-output functions over a range of frequencies (80 to 320 Hz) are presented here; these functions can all be well fit by a hyperbolic tangent curve, similar to the template for receptor response discussed by Lipetz [in Handbook of Sensory Physiology, Vol. I, edited by O. E. Lowenstein (Springer-Verlag, Berlin, 1971)]. Data are also presented that indicate that the dc component increases with respect to the second harmonic component as frequency increases. This result implies that capacitive effects should be included in models of lateral line microphonics. [Supported by NSF and the Deafness Research Foundation.]
Buffeting excitation of structures by fluid flow
Authors: Sevik M. M.
Published: October 2005

Structural vibrations and sound caused by unstable wake flows is of continuing concern. Measurements of unsteady pressures and forces acting on structures of various geometries have been performed during the past decade over a wide range of Reynolds' numbers, Mach numbers, and cavitation numbers. In many instances, the experiments included rigid as well as oscillating bodies. This paper reviews the most significant results obtained on bluff bodies as well as at the trailing edges of streamlined shapes.
Echo suppression in binaural listening
Authors: Zurek P. M. , Leshowitz B.
Published: October 2005

It has been suggested by Koenig [J. Acoust. Soc. Am. 22, 61–62 (1950)] that “a remarkable property of the binaural system is its ability to ‘squelch’ reverberation.” Although echo suppression is a frequently cited capability of the binaural system, its existence has not heretofore been demonstrated objectively. In order to provide such a demonstration, detection experiments were performed using earphones. The subject was asked to detect a delayed noise burst signal in the presence of a leading noise burst masker. The second burst was a delayed replica of the first burst and served as the simulated echo. Both the first burst, corresponding to the direct sound, and the echo were presented binaurally with interaural delays, T1 and T2, respectively. Detection of the echo was measured as a function of the delay of the echo. With a purely diotic presentation (i.e., T1=T2=0) and echo delays less than 10 msec, detection threshold for the echo was approximately 10 dB lower than when the echo was presented with an interaural delay (e.g., T1=0, T2=500 μsec). This result is the opposite of the masking level difference wherein signal threshold is highest for the diotic case.
Lateral position, interaural discrimination, and binaural detection: Model based on auditory-nerve activity
Authors: Stern Richard M.
Published: October 2005

A model for perception of 500-Hz binaural tones featuring a description of their subjective lateral position is presented and compared with lateralization and discrimination data. The model is a continuation of work by H. S. Colburn that proposed a binaural analyzer that interaurally compares auditory-nerve firing times. Subjective position predictions of the present model are obtained by a nonoptimal weighting of these comparisons that depends on the interaural intensity difference of the tone. Discrimination predictions are obtained, assuming that subjective position is used as the sole decision variable and that internal noise is caused by the randomness of the auditory-nerve patterns. The predictions of the model are reasonably consistent with available data on lateralization scaling of 500-Hz pure tones, and the model predicts the form of results of interaural discrimination experiments more accurately than previous models. Predictions for binaural detection experiments are compared with the appropriate data to determine whether performance in binaural detection, as well as discrimination, can be accounted for by a single common decision variable related to subjective lateral position. [Work supported by NIH.]
An auditory-nerve model for interaural time discrimination of high-frequency complex stimuli
Authors: Colburn H. S. , Esquissaud P.
Published: October 2005

Recent data on interaural time discrimination of high-frequency complex stimuli are described by a model based on auditory-nerve activity. This model is a simple extension of a model that was developed for detection and discrimination of low-frequency stimuli and that includes an explicit description of the activity in the auditory-nerve fibers. In the extension to high frequencies it is assumed, consistent with available physiological data, that the high-frequency neurons have instantaneous firing rates that follow the envelope of the tuning-curve filtered signal. Beyond the auditory nerve description, the model assumes a structure for processing timing information on high-frequency fibers that is the same as the structure for low frequencies. [Work supported by NIH.]
Acoustic discrimination within phonetic categories
Authors: Carney Arlene E. , Widin Gregory P.
Published: October 2005

Phonetic categorization implies limits on the discrimination of speech stimuli within phonetic categories. Tests of the validity of this implication should be made with minimal mnemonic requirements, such as same-different tasks. Using the Abramson-Lisker bilabial series varying along the VOT dimension, we obtained both identification and discrimination functions. Discriminations were made using a same-different paradigm; subjects were instructed to respond according to acoustic differences and not phonetic categories. In subsequent feedback trials on a restricted set of stimuli, discrimination performance improved, in a manner indicating that stimuli within categories are not uniformly discriminable. The implication of such results for the definition of categorical perception is discussed.
On the perception of /w, l, y/ and categorical perception
Authors: Frazier Lyn
Published: October 2005

The perception of /w, l, y/ was examined by giving subjects an identification test and a “same-different” discrimination test composed of eleven 380-msec, 3-formant CV stimuli created on a parallel resonance synthesizer. Each stimulus consisted of an initial 90-msec steady state portion, a 60-msec transition, and a 230-msec steady state portion with formant frequencies appropriate to /ε/. The second-formant frequency of the initial steady state was varied in equal steps from 594 to 2995 Hz. The interstimulus interval in the discrimination test was 1 sec .Subjects perceived these stimuli categorically, showing good discrimination near phoneme boundaries and poor discrimination elsewhere. This finding must be interpreted in the light of recent research on the role of phonetic and acoustic memory in an explanation of categorical perception. [Ann. Report Engineering Res. Inst. Univ. Tokyo, 29, 207–214 (1970).]
Acoustic vs phonological factors in vowel identification
Authors: Edman T. R. , Strange W. , Jenkins J. I.
Published: October 2005

Medial vowels and vowels in vowel-/p/ syllables have been shown to be more accurately identified than isolated vowels and vowels in /p/-vowel syllables [W. Strange, J. J. Jenkins, and T. Edman, “Identification of vowels in CV and VC syllables,” J. Acoust. Soc. Am. 58, S59 (A) (1975)]. This result can be accounted for by either phonological or acoustic hypotheses, both of which suggest differential contributions of the initial and final consonants in aiding vowel identification. In the present study we attempted to establish the relative contribution of the acoustic and phonological factors in vowel identification. We compared the identification of nine American English vowels spoken by six adult talkers in four conditions: /b/-vowel, vowel-/b/, /b/-vowel-/b/, and isolated vowels. Error rates were lowest for the /b/-vowel-/b/ syllables (4%) and highest for the isolated vowels (21%). The /b/-vowel and vowel-/b/ conditions generated intermediate error rates (17% and 10%, respectively). These results are compared to the earlier study, and an interpretation emphasizing the acoustic factors of vowel identification is offered.
Effects of vowel context upon labelling the /s/-/š/ continuum
Authors: Hasegawa A. , Daniloff R. G.
Published: October 2005

Although they are true obstruent consonants, fricatives have not been studied as exhaustively as stop consonants. In the following study, we report findings on perception of a set of VC fricative syllables. Specifically, a continuum of fricatives ranging from /s/ to /š/ were synthesized after close examination of natural fricatives. The fricatives were embedded in coarticulated /i/C and /u/C frames. Identification functions, i.e., phoneme labelling, were done for each vowel environment using panels of 10 listeners each. In addition, response time measures and quality ratings were made of the stimuli. In a second study, an ABX paradigm was used to generate discrimination functions. The crossover point for /s/-/š/ labelling, and the magnitude and steepness of discrimination and identification functions were then tested to determine (1) to what degree fricative perception is “categorical” and (2) to what degree labelling boundary depends upon context. Results indicate that there is a substantial context dependency in the labelling crossover point, and that a good case can be made for quasicategorical perception of the /s/-/š/ fricative continuum.
Some ecological constraints on the perception of stops and affricates
Authors: Raphael L. J. , Dorman M. F. , Liberman A. M.
Published: October 2005

Previous research has shown that for a disyllable like /εb dε/, a silent interval between the stop consonants is a necessary condition for the perception of the first, i.e., the syllable-final, stop. Even in the absence of a silent interval, however, the first stop remains intelligible if the first syllable is spoken by a male and the second by a female. Further experiments have confirmed and extended these results. Introducing a silent interval between the carrier phrase, “Now say …” and the word “shop” causes listeners to hear “chop.” If, however, the carrier phrase is spoken by a female, and “shop” by a male the perception of “shop” remains unchanged. Finally, when listeners hear stop consonant sequences produced by a single voice, but reproduced over two loudspeakers separated by 180°, (i.e., /eb/ over one speaker and /de/ over the other), the first syllable remains intelligible even in the absence of a silent interval between the stops. It is as if the listener knew that two vocal tracts, acting independently, can do what a single vocal tract cannot.
Some properties of the auditory component of selective adaptation
Authors: Bailey Peter J.
Published: October 2005

Manipulation of the spectral properties of adapter and test syllables in a selective speech adaptation procedure may provide a means to illuminate the microstructure of that component of the adaptation effect attributed to an auditory level of processing. In a pair of experiments, syllables having formant transitions which varied in rate, duration, and spectral location were synthesized to produce syllable series between different places of articulation and between the categories stop and semivowel. The series of test stimuli were identified in both the unadapted condition and after adaptation with a variety of syllables whose formants shared with the test series one or more of the transitional properties rate, duration, and spectral location. The pattern of phoneme boundary shifts produced by these adapters confirms the importance of spectral similarity between adapter and test syllables as a major determinant of the adaptation effect. The results suggest in particular that spectral similarity can be conveniently considered as spatial contiguity in a neural representation of auditory signals in terms of frequency and time. [Work conducted at The Queen's University of Belfast, Northern Ireland.]