bluuebunny/crossref_metadata_2025 huggingface dataset:
Masking period patterns: presentation of a model consistent with an enlarged set of experimental results
Authors: Schwartz Jean-Luc , Escudier Pierre
Published: October 2005

Masking period patterns (MPP) have been proposed by Zwieker as psychoaoustic equivalent of period histograms (PSTH) measured in auditory neurons. A large set of results have been obtained and represented on a model for predicting MPP [E. Zwicker, Biol. Cybern. 23, 49–60 (1976)]. We have already used this technique for studying PSTH when the basilar membrane is stimulated by trapezoidal patterns of excitation [P. Escudier et al., J. Acoust. Soc. Am. Suppl. 1 67, S102 (1980)]. Our results cannot be completely interpreted on the basis of Zwicker's model. We present a new model, using a peripheral auditory system simulation previously elaborated in our group [J. M. Dolmazon et al., Speech Commun. 1, 52–73 (1982)] and some simple additional hypotheses. This model is shown to be able to predict the whole set of available MPP results.
Duration and rise-decay time in the masking-level difference obtained in pulsed masker conditions
Authors: Yost William A. , Dye R.
Published: October 2005

The MLD (masking-level difference) is small when the signal and masker are pulsed together. The addition of a forward “masker fringe” to the pulsed signal-puls-masker can increase the pulsed MLD to that obtained when the masker is on continuously. The masker fringe must be at least 500 ms in order for the MLD to be as large as that obtained in the continuous masker condition. A parametric study of masker-plus-signal duration in the pulsed conditions and signal rise-decay time in the continuous masker conditions were investigated. Masked thresholds in the NoSo and NoSπ conditions were obtained at 500 Hz from two-point psychometric functions determined in a single-interval procedure. The study was designed to test predictions of the equalization-cancellation model for the MLD and to test the assumption that the small pulsed MLD is due to a lack of change in interaural parameters at signal onset. [Work supported by NSF.]
Effects of masker level on binaural masking patterns
Authors: Koehnke Janet , Cohen Marion F.
Published: October 2005

This study investigated the effects of masker level on the shape of masking patterns for binaural detection. Specifically, detection thresholds of a 700-Hz sinusoidal signal were measured as a function of the center frequency of a narrow-band masking noise. Thresholds for the 350-ms tone were measured twice at four masker levels, once with the signal presented diotically and once with the signal interaurally phase reversed. The masker was a continuous, diotic 80-Hz-wide noise that varied in frequency over a range of four critical bands above and below the signal frequency. Results show that the shape of the masking pattern varies as a function of masker level: with low-level maskers there is a downward spread of masking; with a moderate-level masker, masking is symmetrical; and with a high-level masker there is an upward spread of masking. This dependence is comparable to that observed in monaural masking patterns [E. Zwicker, Audiology 19, 330–334 (1980); E. Zwicker and A. Jaroszewski, J. Acoust. Soc. Am. 71, 1508–1512 (1982)]; this similarity suggests that frequency analysis in the monaural and binaural auditory systems is similar. [Work supported by NIH grant ROI NS16802 and 2T32 NS07099-06.]
Effects of noise and cubic-difference-tone behavior
Authors: Norton Susan J. , Langer Theresa , Jesteadt Walt
Published: October 2005

It has been suggested that external noise maskers, like sensorineural hearing loss, have a linearizing effect on cochlear function and that external noise maskers can be used to simulate sensorineural hearing loss. To test these hypotheses, cubic-difference-tone (CDT) behavior was measured in normal hearing listeners using a forward-masking paradigm for sinusoidal primaries presented in quiet and in the presence of low-pass, high-pass, and broadband noise. The noise was presented only during the primaries and could be considered as an additional forward masker. None of the noises used affected CDT behavior in a way that would support the hypothesis that noise has a linearizing effect on cochlear function. The range over which cubic difference tones could be measured was shifted by the amount that the thresholds for the primaries were shifted, but otherwise CDT behavior was similar to that observed in quiet. In contrast with results reported for persons with high-frequency sensorineural hearing loss, high-pass noise above the primaries had no effect on CDT behavior. Low-pass noise also had no effect on CDT behavior. Broadband noise added with the CDT much like external sinusoids and noise add. [Work supported by NIH.]
The prediction of masking in helicopters
Authors: Patterson R. D. , Lower M. , Wheeler P. , Rood G.
Published: October 2005

The spectrum of the noise in helicopters is complicated because the rotor and blades produce intense low-frequency noise components, and the gears and transmission produce narrow-band peaks in the mid-frequency range. Thus the helicopter provides an excellent proving ground for theoretical models of auditory masking. The rounded exponential model of masking [Patterson et al., J. Acoust. Soc. Am. 57, 1788–1803 (1982)] has been extended to predict threshold at low frequencies and high levels by incorporating the results of recent laboratory studies. Two experiments were then performed to test the model's ability to predict masked audiograms gathered in the presence of helicopter noise. In one experiment the masker was presented via headphones in sound attenuating chambers to 12 listeners, in the other the masker was presented via loudspeakers in a Lynx helicopter simulator to six listeners. The results show that a very simple auditory filter, Roex (p), and mild asymmetry (upper side sharper) can predict the data well.
Point source excitation of cylindrical elastic shells filled with fluid
Authors: Fuller C. R.
Published: October 2005

The vibrations of a cylindrical elastic shell filled with fluid containing a monopole acoustic source are theoretically studied using the spectral equations of motion. The modal response of the shell wall is examined for different radial locations of the source and explained in terms of wave behavior. The distribution of vibrational energy between the shell and the fluid is evaluated at various axial locations and source positions. An interchange of energy of vibration between the shell and the fluid as the wave field propagates along the shell-fluid system is uncovered.
Anomalous recovery of damped radial modes in a circular-sector duct with locally heated flow
Authors: Maham J. R. , Yeh S.-Y.
Published: October 2005

It is often desirable to predict acoustic propagation in a circular duct carrying a locally heated flow. Common examples include jet engines and certain industrial and commercial burners whose combustion-related noise can be an environmental problem if allowed to penetrate into the surroundings. In these cases axial gradients in the steady flow variables, established as a result of local combustion heating of the flowing gas, lead to important modifications of the usual linearized equations expressing conservation of mass, momentum, and energy in the acoustic variables. Thus the classical three-dimensional wave equation no longer applies and a new wave equation must be formulated. In the present work the appropriate modified wave equation is derived for and applied to a duct whose cross section is a sector of a circle of sufficiently small included angle to preclude the existence of circumferential modes in the frequency range of interest. This two-dimensional geometry, though somewhat artificial, has been chosen to allow the first radial mode to be isolated and studied separately. The analysis reveals that the first radial mode, when damped near the inlet to the duct, may under certain circumstances recover and even grow in amplitude as it passes through the heating region. [Work partially supported by NASA under Grant NAG3-124.]
Measuring the transition to turbulence in oscillatory pipe flow
Authors: Manley Martin , Thompson Charles
Published: October 2005

Acoustically induced turbulence, AIT, has recently become a topic of increasing interest. Research in AIT has been directed toward developing processes for aglomerating suspended particles using the nonlinear properties of the host fluid. However, the conditions governing the stability of the oscillatory or Stokes boundary layer have not been investigated. An experimental scheme for investigating the transition to turbulence in oscillatory flows will be presented.
Block encoding of speech spectral principal components
Authors: Holland James R. , Zahorian Stephen A.
Published: October 2005

A Karhunen-Loeve (KL) series expansion was used to block encode speech spectral principal components as a function of time. Each of ten principal components was first obtained as a linear combination of 20 speech spectral band energies. Using a fixed block length of ten frames (0.128 s), the KL basis vectors were computed separately for various speakers for each principal component. However, the optimal KL basis vector set was essentially the same for each principal component and for the different speakers. The basis vector set also closely resembled a cosine basis vector set. Approximately 94% of the variance of the principal components was accounted for by five (out of ten) basis vectors. Speech was synthesized using the KL basis vectors for block encoding of ten-frame blocks of principal components. Informal listening tests indicate that very little information is lost using five basis vectors. These results indicate that speech spectral principal components, particularly the low-ordered ones which reflect the overall spectral shape, are highly correlated in time. [Work supported by NSF.]
On the use of a perceptual distance with speech vector quantization
Authors: Carré René , Lacoste Christian , Tuffelli Denis
Published: October 2005

A vector quantization using perceptual data has been studied. The frequency and bandwidth scales of the poles computed by LPC methods were quantized according to perceptual difference limen. The small sensitivity of the hearing system to the formant amplitude was thus taken into account. A spectrum dictionary was then obtained. One spectrum is a set of quantized poles and the inter-spectrum distance is either 1 or 0. Good results are expected in the case of multi-speaker applications: low bit rate transmission and speech recognition. The method is well adapted for controlling all the steps of the processing and thus for improving the obtained results. For instance, the evolution of the dictionary can be easily studied with an increasing number of speakers and the statistical importance of each spectrum can be measured. First synthesis are quite good and comparable with those obtained by classical vector quantization. A phonetic labelling of the signal will lead to a dictionary spectrum labelling and a pointer giving the number of each successor permits to study the coarticulation effects at the level of the spectra. [Work supported by DRET.]
Optimum coding of formant speech synthesis parameters
Authors: Awad S. S. , Guerin B.
Published: October 2005

An optimum coding of the parameters of a formant speech synthesizer has been found. The optimization procedure is based on statistical and subjective criteria. The synthesizer is a parallel formant synthesizer. The utterances chosen for experimentation are groups of French high-quality synthetic vowel-consonant sounds (voiced stops and voiced fricatives). The proposed procedure consists of three principal steps. The first step is a statistical study on the parameters in order to find the effective range of variation of each parameter. The second step is to determine by subjective evaluatioa the minimum number of bits and type of quantization (linear or logarthmic) needed to quantize each parameter. The third and final step is to optimize the sampling interval depending on the nature of speech events. In fact a variable sampling interval is proposed. We have applied the proposed procedure in order to minimize the total bit-rate necessary for encoding the parameters of the synthesizer and in the same time obtaining synthetic utterances indistinguishable from original utterances obtained from nonoptimized parameters. Finally, throughout this study we have emphasized the different characteristics of each type of the utterances.
The effects of duration adjustments of preceding vowels on fricative voicing perception by hearing-impaired listeners
Authors: Revoile S. G. , Holden-Pitt L. D. , Pickett J. M.
Published: October 2005

For /bæf/, /bæv/, /bæs/, and /bæz/, perception of the fricative voicing distinction was studied for 25 moderately to profoundly hearing-impaired undergraduates. The purpose was to discover whether some listeners might benefit from enhancement of the duration cue in the preceding vowel. Identification was tested for ten utterances of each syllable. These utterances were presented unedited or with their vowels adjusted for duration. The utterances of /bæf/ versus /bæv/ and /bæs/ versus /bæz/ had been selected to differ in the degree of contrast for the vowel duration cue. The unedited utterances best-perceived for fricative voicing contained vowels that were the most salient for the duration cue. The durations of these vowels were exemplars to be approximated among the other utterances that were adjusted for vowel duration. For /bæf/, and /bæs/ utterances, the vowels were shortened through pitch-period deletions; the adjusted /bæv/ and /bæz/ utterances contained vowels lengthened via iterated pitch periods. Preliminary analyses revealed that the utterances with duration-adjusted vowels yielded substantially improved fricative voicing perception for 15 of the listeners. The other listeners showed minimal changes in performance between the unedited versus vowel-adjusted syllables. These were listeners whose perception for the unedited syllables was either very good or very poor.
Spectral tilt and burst duration as determinants of stop consonant identification accuracy for hearing-impaired listeners
Authors: Dorman Michael N. , Lindholm Julie M. , Hannley Maureen
Published: October 2005

In a series of experiments we have manipulated the tilt of the onset spectrum and duration of the release burst in syllables with initial voiced stop consonants. In identification tests we have found that normal hearing listeners are largely unaffected by abnormal tilt or burst duration. Hearing-impaired listeners, on the other hand, are significantly affected. We speculate that when moderately and severely hearing-impaired listeners achieve normal recognition accuracy for stop consonants, they do so by giving greater-than-normal perceptual weight to the information provided by burst duration and spectral tilt. The results of identification tests in listeners with congenital severe to profound sensorineural hearing impairments are consistent with our speculation.
Acoustic-phonetic analysis of Japanese
Authors: Aoki Chieko , Klatt Dennis , Kawasaki Haruko
Published: October 2005

How difficult would it be to convert a synthesis by rule program for English (DECtalk) to speak Japanese? What is the ideal, hopefully minimal, corpus of recorded materials that would have to be spectrally analyzed in order to derive sufficient information for a first order approximation to Japanese? We have recorded three male speakers reading a list of a little over 100 nonsense syllables, which exhaust the CV(C) syllable inventory of Japanese, and provide information about segmental contrasts. One of the speakers then read a 500-word text to provide information on sentence-level phenomena. We have measured segmental durations, formant motions, and deletion phenomena in broadband spectrograms of this recording, and are beginning to formulate rules for synthesis on the basis of computer analyses of the data. The approach now appears far less unreasonable than it did when we began. Some results of the analysis will be presented. [Work supported in part by an NIH grant.]
Identifying familiar talkers over a 2.4 kbps LPC voice system
Authors: Schmidt-Nielsen Astrid
Published: October 2005

A commonly cited drawback of narrow-band digital voice communication systems such as the DoD standard LPC algorithm is that talker recognition is poor. Yet is is the opinion of many users that they frequently recognize the talker. Tape recordings of 24 talkers conversing over an unprocessed channel and over an LPC voice processing system were subjected to listening tests. The listeners were 24 co-workers who listened to the tapes and attempted to identify each talker from a group of about 40 people working in the same branch. Prior to the recognition test, each of the listeners also rated his or her familiarity with each of the talkers and the distinctiveness of each talker's voice. There was some loss in voice recognition over LPC, but the recognition rate was still quite high. Unprocessed voices were correctly identified 88% of the time, whereas the same people talking over the LPC system were correctly identified 69% of the time. Talker familiarity was significantly correlated with correct identifications. There was no significant correlation between the rated distinctiveness of the talker and correct identifications. However, familiarity and distinctiveness ratings were highly correlated. This suggests that people consider a familiar voice to be distinctive regardless of other characteristics that might make that particular voice stand out in a crowd. [Work supported by NAVELEX.]
Pilot preferences, intelligibility, and learnability of two types of computer generated speech
Authors: Simpson Carol A. , Navarro Teresa
Published: October 2005

Two types of computer speech were compared to obtain helicopter pilots' preference judgments and to determine if intelligibility, learnability, and/or confidence ratings would vary as a function of speech type, sex of listener, or whether the listener was a pilot. The two speech types were direct synthesized speech—generated by a Votrax ML-1 synthesizer, and an LPC-encoded version of the direct synthesized speech using the TMS-5220 speech chip. Stimuli for the preference study were sentence length helicopter voice warnings. Stimulus material for the second study was a PB word list. The direct synthesized speech was preferred by the pilots in the first study and, in the second study, was significantly more intelligible than the LPC-encuded synthesized speech. A significant interaction between speech type and runs indicated that intelligibility improved with exposure to PB words for direct synthesized speech while there was no improvement for LPC-encoded synthesized speech. Confidence ratings agreed with the intelligibility data. Spectrograms of the two speech types reveal differences which might explain the differential learnability. [Funded by NASA-Ames.]
A method of compensating for missing elements in uniform linear arrays of sensors
Authors: Stockhausen John H. , Farrell Joseph B.
Published: October 2005

This paper presents a new technique for reducing the performance degradation due to nonfunctional or overly noisy elements in a uniform line array. Performance degradation appears as a loss of field mapping accuracy, and measures taken to recover this lost accuracy often lead to increased variance in the estimates of beampower. Thus the degree of time averaging required to reduce random errors is increased. The new technique uses data from all of the properly functioning sensors to estimate the complex Fourier coefficients (excitations) which would have been present at the locations of the degraded elements. This procedure can then be followed by conventional filled-array beamforming. In terms of bias and random errors in predicted beampowers, the procedure yields results comparable to several more familiar compensation techniques for nearly isotropic fields. It exhibits a definite advantage relative to the other techniques, however, for fields which are highly anisotropic.
Energy density signal analysis on the time-frequency plane
Authors: Mohammed A.
Published: October 2005

To analyze time-varying signals, a modification of the conventional FFT technique is introduced to make use of the phase information that is normally ignored. With this modification, the resulting display gives the energy density as a function of the local time and frequency attributes of signals and, at the same time, achieves a resolution higher than that of the conventional method. Though this modification is a geometric transformation of the conventional spectrum that was investigated earlier [K. Kodera et al., Phys. Earth Planet. Int. 12, 142–150 (1976)], this presentation emphasizes the rationale behind the use of local signal attributes which are fundamental concepts of signal propagation in terms of wave packets. More important, the method of implementation is unique with a computational accuracy and simplicity that allows for analysis over a large dynamic range and for an online real-time capability. Included are examples of results for time-varying signals both without and with strong noise backgrounds as well as a comparison with results obtained by the conventional FFT technique.
Source depth estimation in waveguide
Authors: Shang E. C.
Published: October 2005

In the present paper a high resolution approach of source depth estimation in waveguide has been proposed. The method was based on the mode filtering technique. It was well known that a high resolution estimator can be obtained by designing a vector orthogonal to the target signal component contained in the data vector, such as the maximum entropy method. By using a mode filter, the field sampling data of a vertical array was transformed to a vector which consists of two parts: the eigenfunction of each mode and the modal phase corresponds to the source range. The latter part can be compensated by means of range information extraction proposed in our previous paper [Shang et al., J. Acoust. Soc. Am. Suppl. 1 74, S78 (1983)]. Then the data vector consisting of a finite number of mode eigenfunctions becomes an “almost orthogonal” and can be used to construct a high resolution power processor.
Eigenvalue methods for bearing estimation algorithms
Authors: Johnson Don H.
Published: October 2005

Many high-resolution bearing estimation algorithms involve the computations of the spatial correlation matrix. The utility of applying an eigenanalysis to the spatial correlation matrix has been described by many workers for a variety of algorithms. A cohesive theory for analyzing the properties of these algorithms is presented. The accuracy of bearing estimates is shown to be directly related to the time-bandwidth product of the computations. However, the various algorithms depend on this quantity in different ways. Adaptive beamforming is shown to yield somewhat better estimates than linear predictive methods when eigenanalysis is used. [Work supported by ONR.]
The formation of a synthetic aperture with towed hydrophones
Authors: Carey W. , Yen N.
Published: October 2005

Passive-synthetic apertures were formed with experimental data from the coherent summation of subapertures formed from 1–4 hydrophone groups at successive time samples. The results show that the coherent signal gain for synthetic apertures with lengths up to 95 λ was a function of the number of successive time samples (Nts) described by α 10 log(Nts) with α∼0.7. The hydrophones were towed with a speed-to-wavelength ratio of 0.168 s−1 at a depth of 26 λ. A calibrated moored source was at a depth of 37 λ in 3200 m of water at distance-to-wavelength ratio of between 20 and 30 × 103 from the receiver. The stable environment was characterized by a sound-speed profile with a negative gradient of − 0.130 s−1 to a sound channel axis of 150 m and a positive gradient of + 0.016 s−1 to a critical depth of 1825 m. The resulting propagation was RR and RSR with losses of between 90 and 100 dB. These results show that for these environmental-acoustic conditions the synthetic apertures can be formed by the coherent summation of phase corrected subaperture beams over successive time samples as long as the synthetic aperture length is less than the single path coherence length and the synthetic aperture processing time is less than the signal's temporal coherence length. The results presented represent an extension of earlier work [R. Fitzgerald, J. Acoust. Soc. Am. 60, 752-753 (1976); R. Williams, J. Acoust. Soc. Am. 60, 60–73 (1976)] by the demonstration that coherent gain can actually be achieved with resolution such that multipath vertical arrival angle differences can be resolved.
Receiver directivity modeling in three dimensions
Authors: Tappert Fred , Nghiem-Phu Lan
Published: October 2005

A technique has been developed to compute the quantity that is represented in the passive sonar equation as TL′ = TL − AGs, where TL is transmission loss and AGs is receiver signal array gain. In full-wave theory these two quantities are not logarithmically additive because beam deviation loss is not separable, and hence TL′ must be computed directly. It is required to compute TL′ as a function of source coordinates x,y,z for given receiver location, given frequency, and given farfield beam pattern (depending on steering angle). Making use of reciprocity, we have used our PE model to compute TL′ in realistic range-dependent ocean environments. Results are displayed as contours of TL′ in the vertical x,z plane at various azimuths, and as contours of TL′ in the horizontal x,y plane at various depths.
Vertical directionality and depth dependence of wind-generated noise
Authors: Renner William W. , Kirby William D.
Published: October 2005

For highly directive sensors, the spatial coherence of the background noise can be an important consideration in estimating system performance. An isotropic vertical-noise field, as often assumed, is typically an unrealistic characterization of the true field. In this paper we present a vertical-noise model for high-frequency, wind-generated noise based on the earlier work of Talham [J. Acoust. Soc. Am. 36, 1541–1544 (1964)] extended to include the volume absorption model more recently developed by Fisher and Simmons [J. Acoust. Soc. Am. 62, 558–564 (1977)]. The impact of various environmental factors upon both the directivity and the depth dependence of high-frequency noise is addressed. Limitations of existing boundary loss models and their implications for noise modeling are discussed.
Generation of uniform progressive periodic plane waves within a volume near an array of randomly positioned point sources
Authors: Moses Bruce O.
Published: October 2005

A method is presented for computing complex shading coefficients for a specified array of randomly positioned point sources to produce an arbitrarily designed steady-state uniform progressive periodic plane wave. The coefficients are designed to produce the plane wave within a specified volume in the nearfield of the projecting array. The coefficients have been modeled on a computer, and contour plots of the resulting pressure field have been produced. The direction of propagation of the plane wave can be controlled by changing the shading coefficients for the elements of the array. The complete farfield equivalent beam pattern of an unknown transducer positioned within the controlled volume can be accurately measured while holding both the unknown transducer and the array fixed in position without rotation. The method opens many possibilities for improved acoustical calibration techniques.
Development of a new underwater acoustic test and calibration center
Authors: Zets G. A.
Published: October 2005

A new underwater acoustics test and calibration center has recently been established in Cape Town, South Africa. With the 2.5 million liter main test tank measuring 20.2 × 11.2 × 11.2 m and a second tank measuring 20.7 × 3.5 × 4.2 m, the facility is amongst the largest in use today. Designed to meet the requirements of users involved in military, industrial, and research activities, the facilities incorporate some special features such as vibration isolation mounting of the tanks and the capability of supporting marine life for fisheries research. This paper presents details of this new facility and illustrates some interesting aspects of the design and construction of the tanks. The choice of instrumentation and signal processing techniques will be discussed as well as proposed methods for handling and positioning large transducers.
Underwater-nearfield acoustical holography using a robotic controlled scanner
Authors: Williams Earl G. , Dardy Henry D. , Fink Richard G.
Published: October 2005

A computerized, experimental facility has been developed to implement nearfield acoustical holography in a large underwater tank. The facility contains a robotic controlled hydrophone arm which allows the precise positioning and motion of a hydrophone in the tank completely under computer control. Automated scanning is done close to,the surface of a radiating/scattering object and the resulting pressure data are stored digitally. The frequency range of operation is dc to above 30 kHz. Using the principles of nearfield acoustical holography [Williams and Maynard, Phys. Rev. Lett. 45, 554 (1980)] this data are forward and backprojected to study the radiated/scattered acoustic fields. Forward projection provides directivity patterns, and intensity and vector-velocity field mappings. Backward projection provides surface velocity and pressure (fluid loading)—reconstructing the field in the extreme nearfield. Experimental results are shown providing a complete description of the field radiated underwater from a point-driven rectangular plate. Shown are its surface velocity, intensity vectors in the extreme near field, fluid loading effects. radiation efficiency, and directivity patterns.
Range effects for speech and nonspeech Judgments of sine wave stimuli
Authors: Studdert-Kennedy Michael , Williams David R.
Published: October 2005

In a previous paper [Williams et al., J. Acoust. Soc. Am. Suppl. 1 74, S66 (1983)], we demonstrated that an influence of consonantal context on vowel perception [B. Lindblom and M. Studdert-Kennedy, J. Acoust. Soc. Am. 42, 830–843 (1967)] is obtained for sine wave as well as for formant stimuli. We showed further that a similar influence is not apparent when the sine wave stimuli were judged in terms of pitch rather than speech categories. Category boundaries for pitch judgments fell at the midpoint of the continua while those for isolated vowels and vowels in consonantal context were arrayed to the left and right of the midpoint, respectively. In the present set of experiments, we sought to assess the stability of the speech and nonspeech category boundaries by testing an effect of stimulus range on listeners' categorizations. To do this, we removed the lower one-third from each of the continua used in our previous experiment. Pilot results replicate our earlier findings: once again, pitch boundaries divided the continua in half and speech boundaries fell to either side of the midpoint. Nonetheless, these data show that speech as well as pitch judgments are subject to range effects. [Work supported by NICHD.]
Formant frequency lowering during acquisition of the French/r/
Authors: Manning Walter H.
Published: October 2005

Four adult males, all native speakers of English, were trained to produce the French /r/ during ten training sessions. Formant frequency values for /r/ were spectrographically analyzed based on the subjects' production of ten test words in each training session. In order to observe the importance of auditory feedback during the acquisition of /r/, subjects produced the ten test words under conditions of no masking and 100 dB (SPL) binaural competing speech masking. Formant values were plotted as F1−F2 and F2−F3 relationships. A decrease in formant frequencies across training was observed during both unmasked and masked conditions. F2 values showed the greatest change across training. Formant values were consistently lower during the masked conditions. One explanation for the decrease in formant values appeared to be an increase in tongue height combined with a lowering of the velum. Results are contrasted with perceptual judgments of /r/ across training.
Perception of “modulation” in speechlike signal III: A psychoacoustic or phonetic dimension?
Authors: Porter R. J. , Miller C. J.
Published: October 2005

We have proposed that a “modulation sensation” may accompany second-formant variations like those found in speech [Porter and Miller, J. Acoust. Soc. Am. Suppl. 1 73, S3 (1983); Miller and Porter, J. Acoust. Soc. Am. Suppl. 1 74, S66 (1983)]. Unlike some other psychoacoustic attributes of formant variations, modulation sensations appear to be preserved when the transitions are presented either within or without first formant context. Modulation might, therefore, serve as a psychoacoustic basis for some phonetic decisions. On the other hand, the modulation sensation might be as phonetically irrelevant as, for example, a signal's loudness. To examine these alternatives, single haversine-cycle modulations of second formants were presented in the context of an F1. Systematic variations in the direction and extent of F2 modulation were coupled with variations in the amount of a simultaneous haversine modulation of F1 amplitude. The resulting set of signals define an acoustic space which casual listening suggests might be perceptually divided along phonetic dimensions as well as along psychoacoustic dimensions of modulation magnitude and quality. Comparisons between listeners' judgments of modulation magnitude, discrimination of signals, and phonetic labeling will be presented and discussed in terms of the alternatives presented above. [Supported in part by The Lousiana Eye and Ear Foundation, The Lousiana Lions, and the Kresge Foundation.]
VOT, /s/-noise duration and silent interval as trading cues; stops after /s/
Authors: McCasland George P.
Published: October 2005

In the results of the writer's previous perceptual experiment [Phonetica 34, 218–228 (1977)] which was based on stimulus materials edited from spoken utterances, the minimally contrasting test utterances it's dill, it still, it's still were perceptually distinguished by appropriate values of /s/-noise duration and silent interval in the context of short delayed VOT. The utterance it's till containing a voiceless aspirated stop after /s/ was heard in 75% of the judgments of otherwise similar stimuli containing long delayed VOT. A recent reanalysis of the other 25% of the judgments suggests that the potency of the long delayed VOT in these tokens had been weakened in trading with /s/-noise duration and silent interval. These findings have been replicated in perceptual data from a new group of subjects. Lisker's work [Haskins Final Report 11, B-11, 1–9 (1965)] based on spectrographic measurements of similar utterances has been an important resource. For the stop after /s/ at word boundary, VOT as a voicing cue appears to be involved in a three-way trading relationship.
Effects of temporal stimulus properties on perception of the [sl]-[spl] distinction
Authors: Repp Bruno H.
Published: October 2005

Two studies investigated the independent influences of the durations of preceding and following signal portions on the amount of closure silence needed to perceive “splash” rather than “slash.” The durations of the [s] and [l] acoustic segments had opposite effects which canceled when the silence was short (Exp. 1) but yielded a net effect due to [s] duration when the silence was long (Exp. 2). These findings, which resolve a conflict between earlier results in the literature, are interpreted as reflecting a perceptual compensation for coarticulatory shortening of [s] before stop consonants, in conjunction with (possibly psychoacoustic) contrastive interactions between the perceived durations of adjacent acoustic segments. [Research supported by NICHD.]
Influence of following context on perception of the voiced—voiceless distinction in syllable-final stop consonants
Authors: Williams David R. , Repp Bruno H.
Published: October 2005

The duration of the vowel preceding a syllable-final stop consonant (as defined by some acoustic criterion) generally is a potent cue to the perception of the stop as phonologically voiced or voiceless. We show that the absolute vowel duration that characterizes the voiced—voiceless boundary in a monosyllable is reduced considerably when a second syllable beginning with a different stop consonant is added to form a disyllabic word. We also show that this contextual effect declines with increasing temporal separation between the two syllables, and we examine this decline in relation to listener's judgments of whether they hear the stimulus as one disyllabic word or as two successive monosyllabic words. The effect of syllabic context, apart from introducing closure duration as an additional voicing cue, may reflect listeners' tacit knowledge of the temporal compression that occurs in speech production as the number of syllables in a word is increased. [Research supported by NICHD.]
Are selective adaptation and contrast effects really distinct?
Authors: Diehl Randy L. , Kluender Keith R. , Parker Ellen M.
Published: October 2005

Despite evidence [e.g., R. L. Diehl et al., J. Exp. Psych: Hum. Percept. Perform. 6, 24–44 (1980)] that selective adaption and contrast effects are produced by the same mechanisms, Sawusch and Jusczyk [J. Exp. Psych: Hum. Percept. Perform. 7, 408–421 (1981)] reported a dissociation between the effects and concluded that adaptation and contrast occur of separate processing levels. Specifically, they found that a VOT test stimulus near the [b]/[p] boundary was more likely to be labeled “b” following adaptation with [ph a] and more likely to be labeled “p” following adaptation with [ba] or [spa] (the latter consisting of [ba] preceded by [s] noise). In the contrast session, where the long adaptation sequence was replaced by a single context stimulus occurring before or after a single test item, the [ba] and [ph a] contexts had contrastive effects similar to those of the [ba] and [ph a] adaptors, but the [spa] context produced an increase in “b” responses to the test stimulus, an effect opposite to that of the [spa] adaptor. One interpretation of this difference is that the rapid repetitive presentation of the [spa] adaptor gave rise to “streaming,” whereby the [s] was perceptually segregated from lower-frequency [ba] component. In our experiment, we first replicated the results of Sawusch and Jusczyk, using procedures similar to theirs. Next we increased the interadaptor interval to remove the likelihood of stream segregation and found that the adaptation and contrast effects converged. [Work supported by NINCDS.]
Revisiting stop-consonant perception for two-formant stimuli
Authors: Ohde Ralph N. , Stevens Kenneth N.
Published: October 2005

The aim of this study was to re-examine the factors leading to stop consonant perception for consonant-vowel stimuli with just two formants over a range of vowels, using stimuli and test procedures somewhat different from those used in earlier studies. Five two-formant CV stimulus continua were synthesized, each covering a range of second-formant (F2) starting frequencies, for vowels corresponding roughly to [iiæɑu]. Listeners identified the beginning of the syllable under two different instructions: a forced-choice response of b, d, or g, and an open response set. The stimuli with the most unanimous responses were those with substantial F2 transitions, i.e., [bi bɪ du]. There were few consistent g responses. Under open response instructions, stimuli with straight or small transition ranges were identified as beginning with no consonant. The results are compared with those of other studies, particularly with regard to differences in stimulus characteristics, and are discussed in relation to theories of invariance. [Supported in part by grants from NIH.]
Coarticulation effects on fricative consonants across languages
Authors: Nartey Jonas N. A.
Published: October 2005

Previous studies (e.g., Nartey, 1981, 1982, and Soli, 1981) indicate that there is reliable vowel coarticulation effects on fricatives produced in the environments of [i], [a], and [u]. The present study further explores the notion of vowel coarticulation effects On fricatives both within and between languages. Critical band analysis was performed on the fricative consonants of 12 genetically diverse natural languages. These were recorded in the intervocalic environments of the three peripheral vowels i-i, a-a, and u-u. The resulting auditory spectra were compared for fricatives both within and between languages. Results indicate, among other things, that coarticulation rules are language specific. Furthermore, it appears that synchronic coarticulation rules can serve as a window by which we can observe dinchronic sound changes such as Proto Slavic /x/ to modern Polish /∫/.
More on the /t/:/d/ distinction in American alveolar flaps
Authors: Scott Donia R.
Published: October 2005

In American English, alveolar stop consonants occurring in intervocalic position tend to be articulated as a flap (or tap). It has long been claimed that the duration of the vowel immediately preceding the flap is a primary cue to its underlying voicing status. Evidence will be presented to show that listeners can discriminate between naturally produced minimal pairs containing flaps (e.g., coating, coding) based on cues other than preceding vowel duration or the acoustic properties of the flap itself. Consistent differences in phonetic quality and overall quantity between such word pairs will be discussed.
Memory modes and perceptual anchors in categorical perception
Authors: Macmillan Neil A.
Published: October 2005

A theory of intensity perception developed by Durlach and Braida [J. Acoust. Sac. Am. 46, 372–383 (1969), et seq.] is applied to data from categorical perception experiments. According to the theory, two memory modes are used in processing perceptual continua. In the trace mode, observers compare stimuli with the memory traces of other stimuli, and performance is limited by the inter-stimulus interval. In the context mode, observers compare stimuli to perceptual anchors, and performance is limited by the stimulus range. Analysis of existing speech and nonspeech data reveals that stimulus domains differ in (a) the amount of trace variance, (b) the amount of context-coding variance, and (c) the existence and location of anchors; but no single parameter captures the categorical/continuous distinction. Memory variances and anchor locations can be estimated from experiments in which fixed-level discrimination, as well as identification and roving-level discrimination, is measured. Among the few experiments with categorically perceived continua that have used the critical fixed-level condition are some in which discrimination peaks arise from anchors, and others in which they reflect regions of high basic sensitivity. [Work supported by NIH.]
Passive isolation of industrial shock
Authors: Andrews Francis J.
Published: October 2005

Passive shock isolation can frequently be analyzed using a shock nomogram. However, for nonlinear isolation systems, this approach will result in low estimates of transmitted force for a stitching system or low estimates of dynamic deflection for a softening (buckling) system. In such cases, energy methods are used to improve the accuracy of estimated performance characteristics. This presentation will discuss use of the shock nomogram and energy methods for each isolator type, as well as performance advantages of each isolator type. Shock test results will be included.
Flexural vibration of printing cylinders on rotary web offset printing presses
Authors: Evans John W.
Published: October 2005

Analytical and experimental studies on a flexural vibration problem characteristic of rotary web offset printing presses are described. This problem results in a printing defect referred to in the industry as “streaking.” Applications of finite element analysis and vibration measuring instrumentation system to reduce vibration levels are described.
Noise propagated into a fluid by sliding contacts
Authors: Tucchio Michael A.
Published: October 2005

The noise propagated into a fluid associated with the relative motion and rubbing of two surfaces, one of which is in contact with the fluid, is investigated. The method presented in this report uses the finite element method to account for the fluid-structure interaction. The friction forces are calculated using the nonlinear equations of motion as derived from rigid body mechanics. Verification of the technique is provided with an experiment. A weight is abruptly moved over the surface of a simply supported beam which is in contact with the water. The pressure-time response of the resulting noise in the water is compared to the calculated pressure-time response from the finite element method. The correction applied to the plane strain finite element solution to convert it to spherical spreading in a semi-infinite medium uses the relationship between cylindrical and spherical spreading in an infinite medium. This area is in need of additional work. [Research supported by NUSC.]
Physiology, morphology, and pharmacology of neurons in the anteroventral cochlear nucleus of the mouse studied in brain slices
Authors: Oertel Donata , Wu Shu Hui
Published: October 2005

Intracellular recordings from brain slice preparations of the anteroventral cochlear nucleus show that the two major cell types, bushy and stellate cells, have different intrinsic membrane properties. To correlate physiological properties with morphology, cells were injected with horseradish peroxidase after their electrical properties were studied. Bushy cells have nonlinear current-voltage relationships around the resting potential; stellate cells have linear current-voltage relationships. Both bushy and stellate cells respond to electrical stimulation of the auditory nerve with early excitatory and later inhibitory postsynaptic potentials. The properties of bushy cells allow them to preserve the temporal firing pattern of excitatory synaptic inputs. To learn what neurotransmitters might mediate synaptic inputs onto stellate and bushy cells, bath applications were made of presumptive neurotransmitters and their antagonists. Inhibitory synaptic responses to stimulation of the auditory nerve stump are blocked by 10−6 M strychnine. The input resistance of cells dropped in the presence of glycine and gamma-aminobutyric acid at 0.1 to 5 mM. All cells tested were completely insensitive to L-glutamate and L-aspartate at concentrations up to 10 mM. [Work was supported by NIH, NS 17590.]
Responses to amplitude modulation in the cochlear nucleus: A hierarchy of enhancement
Authors: Frisina Robert D. , Smith Robert L. , Chamberlain Steven C.
Published: October 2005

Dynamic changes in sound amplitude are prominent features of naturally occurring auditory stimuli. The neural responses to amplitude-modulated sounds may play important roles in speech encoding, pitch perception, and high-frequency sound localization. Here we report the results of a single unit study of the cochlear nucleus and auditory nerve of the anesthetized gerbil. We find that units of the ventral cochlear nucleus exhibit an enhanced response to amplitude modulation (AM) relative to units of the auditory nerve. This confirms and extends Møller's [Acta Physiol. Scand. 98, 157–167 (1976)] findings in the rat cochlear nucleus. Our results indicate that the amount of response enhancement is commensurate with a unit's rank in a hierarchy based on departure from a primarylike response. Onset units show the strongest phase locking to AM, followed by chopper, primarylike-with-notch, and primarylike units, respectively. A comparison of three-dimensional plots of response modulation versus average intensity and modulation frequency illustrates that the enhancement occurs over a range of AM frequencies between 40 and 1000 Hz, at intensities up to 90 dB above a unit's threshold.