bluuebunny/crossref_metadata_2025 huggingface dataset:
Two-tone suppression in hearing-impaired listeners
Authors: Kramer M. B. , Wightman F. L.
Published: October 2005

Forward-masked thresholds were measured in three listeners with precipitous high frequency losses attributed to cochlear damage. The signal was a sinusoid; the masker consisted of two sinusoidal components: one fixed at the frequency of the signal, and one at a variable frequency. It has recently been reported that with the variable masker at certain frequencies and intensities, signal threshold is lower than with this additional masker component absent [R. V. Shannon, J. Acoust. Soc. Am. 59, 1460–1470 (1976}]. This decrease in signal threshold is termed unmasking and may be interpreted as resulting from suppression of the fixed masker by the variable masker, thus reducing its masking effectiveness on the signal. This unmasking effect was not observed in hearing-impaired listeners if the variable masker was in the frequency region of the hearing loss. In normal frequency regions, unmasking effects were similar to those observed in normal listeners. This suggests that suppression occurs at a level of the auditory system higher than the basilar membrane, possibly at the hair cell level. [Work supported in part by NTH Grant NS12045.]
Temporary threshold shift and exposure noise bandwidth
Authors: Malott K. , Stebbins W. C. , Moody D. B.
Published: October 2005

Audiograms were obtained from three Old World monkeys by positive-reinforcement operant-conditioning procedures. Three series of noise exposures were administered in which the width of the exposure band was varied between 0.15 and 4 octaves, with overall level held constant within each series at 100–108 dB/SPL. In two series, the upper cutoff of the exposure band was held constant at 2 and 4 kHz respectively. In the third series, the center frequency of the exposure band was held constant at 2 kHz. When exposure noise bandwidth was 0.15 octave, the maximum of the threshold shift versus frequency function was at or above the upper cutoff of the exposure band. When the bandwidth was increased with the upper cutoff held constant, the maximum of the function frequently shifted toward lower frequencies and the proportion of the area under the curve below the upper cutoff increased. When the bandwidth was increased with the center frequency held constant, no consistent shifts in either direction were observed in two subjects; for the other subject the maximum shifted toward higher frequencies. The implications of these findings and their relation to the human literature will be discussed. [Supported by Research Grant NS05077 and Program Project Grant NS 05785 from NINCDS.]
Auditory effects of 24-h exposures to noise
Authors: Mills J. H. , Gilbert R. M. , Adkins Warren Y.
Published: October 2005

Groups of human observers were placed in a diffuse sound field for 26-h and exposed to an octave-band noise centered at 4.0 kHz. Levels of the noise were 75, 80, 83, and 88 dB SPL. Threshold shifts produced by the noise increased for the first 4–8 h of exposure and then reached a plateau or asymptote. Threshold shifts at asymptote for test tones of 4.0 or 6.0 kHz increased about 1.7–2.0 dB for every 1-dB increase in the level of the noise above about 72 dB. However, at the frequency of maximum threshold shift (4.5–5.5 kHz depending upon the observer), the increase in threshold shift was about 3.0 dB for every 1-dB increase in the level of the noise above 73 dB. Complete recovery of auditory thresholds occured for every observer and required between 16 and 48 h. Such long recovery times were observed even when the magnitude of the threshold shift at asymptote was less than 10 dB. [Supported by NIEHS.]
Aural nonlinearity and auditory fatigue
Authors: Cobb James P. , Erdreich John
Published: October 2005

Differences in individual susceptibility to auditory fatigue have been ascribed to factors relating to cochlear bio-chemistry, changes in neural excitability, hair cell damage and overdriving of the transducer organ as evidenced by nonlinear distortion. In an investigation of the relation between fatigue and auditory nonlinearity we examined two aspects of temporary threshold shift (TTS): (1) growth of TTS with level of the fatiguing stimulus and (2) the absolute level of TTS for a single fatiguing stimulus amplitude. Nonlinear distortion levels for the listeners were estimated using the tone-on-tone masking paradigm. If individual susceptibility to TTS is determined by factors related to maintenance of the physiological environment in the cochlea, then these factors might mediate absolute levels of TTS in response to a fixed intensity stimulus. In this instance the rate of change of TTS with fatiguing level could be a function of the increased overloading between fatiguing levels. Conversely, if TTS susceptibility is a function the degree of overloading, then absolute levels of TTS should be proportional to nonlinear distortion. The results of this study support a relation between absolute TTS levels and nonlinear distortion.
Auditory and vestibular function in human subjects exposed to oxygen at two atmospheres absolute
Authors: Neff P. A. , Hendricks P. L. , Hunter W. L. , Rensink M. J. , Morioka W. T. , Cantrell R. W.
Published: October 2005

Experiments were conducted to test the hypothesis that intermittent exposure to oxygen at two atmospheres absolute (2 ATA) could delay the onset of pulmonary toxicity. This paper reports effects of 2 ATA O2 (total exposure time of 14.1–19.3 h) upon the auditory and vestibular systems of five diver volunteers and effects of a normoxic N2O2 mix on a sixth. Immediately after exposure four subjects had increased acuity by air conduction in the mid-frequency range. Bone conduction sensitivity increased for all six subjects. Middie-ear reflexes were depressed or absent with only partial recovery one month after the experiment for the O2-breathing subjects, but were more sensitive after exposure for the subject breathing N2O2. Middle-ear pressures grew markedly more variable for O2 subjects up to 72 h postexposure, implying decreased eustachian tube function from oxygen. Vestibular systems of four O2-exposed subjects and the N2O2 subject were markedly suppressed to caloric stimulation after exposure, implying pressure as the contributory cause. Operational application may be found in possibly sensitizing certain frequencies to improve the listener's ability to detect signals in noise. [Work supported by CIC, NRMC. and COMSUBDEVGRU ONE.]
Improvement and evaluation of the vibration pick-up type ear microphone
Authors: Ono Hiroshi , Saito Shigeji
Published: October 2005

We developed a vibration pick-up-type ear microphone which was formerly reported at the 90th ASA Meeting. After evaluating its function and comparing it with other conventional microphones, we concluded as follows; It is possible to obtain remarkable sensibility without closing the ear canal and to obtain high intelligibility voice in level over 95% of sound articulation. There was no significant difference between voice intelligibility with a mask and that without a mask. Sound articulation in high ambient noise of the vibration pick-up type ear microphone (110 dB—96%, 125 dB—84%) is much higher than that of noise cancelling microphone (110 dB—87%, 125 dB—77%) and sound pressure type ear microphoen with ear muff (110 dB—85%, 125 dB—68%). By use of the earphone-microphone, we can simultaneously talk and hear through the ear, thanks to no necessity of obstruction of the ear canal. Applications of the vibration pick-up type ear microphone are (1) howlingless microphone, (2) free hand microphone, (3) mask microphone, and (4) microphone for voice communication in high ambient noise. We have made communication systems with the vibration pick-up-type ear microphone and evaluated in various environments.
Automatic detection of nasal/non-nasal feature in continuous speech
Authors: Li K. P.
Published: October 2005

Nasals usually have wider-formant bandwidth than vowels and other nonvowel sonorants. In an LPC analysis, if the last reflection coefficient is set near to unity, then all poles of the inverse filter in the z plane are also shifted near to the unit circle; however, those poles do not follow the radial direction. Therefore, formant frequencies found by shifted spectrum show differences from corresponding formants derived by original unshifted poles. The larger the bandwidth of the pole is, the larger is the difference between the original formant frequency and the shifted formant frequency. Nasals show particular large differences of the first and second formants. An experiment on continuous speech to classify nonvowel sonorants to nasal or non-nasal has been tested. Results are discussed.
Evaluation of a voiced-unvoiced-silence analysis method for telephone-quality speech
Authors: Schmidt C. E. , Rabiner L. R. , Atal B. S.
Published: October 2005

Recently a statistical decision approach to the problem of voiced-unvoiced-silence detection of speech was proposed by Atal and Rabiner [IEEE Trans. on Acoust. Speech Signal Process., June (1976)]. This method was found to perform well on high-quality speech. However, the five speech parameters used in the analysis were not found to be as good for telephone-quality speech. Thus an investigation was undertaken to determine the five best parameters to use for a voiced-un-a reliable voiced-unvoiced-silence decision across a variety of standard telephone connections. A large number of parameters (69) were included in the investigation. including 12 LPC coefficients, 12 correlation coefficients, 12 PARCOR coefficients, 12 LPC partial-error terms, etc. Many of the parameters were immediately eliminated because they provided almost no separability between the three decision classes. The remaining parameters were used in a knock out optimization to determine the five best parametes to use for a voiced-unvoiced-silence analysis. Various error weights were investigated to see what types of errors occurred and how they could be minimized. Finally, the use of the Itakura two-pole normalization was investigated to see its affect on the error scores.
Real-time harmonic pitch detector
Authors: Seneff S.
Published: October 2005

A new pitch extraction algorithm has been developed which is designed to be robust against the degradations encountered in telephone speech. The algorithm determines the pitch from the spacing between harmonics in the first formant region, using an iterative technique whereby larger peaks are weighted more heavily. The desired spectral region is obtained with better than 7 Hz resolution using only a 128-point FFT, after simple digital signal processing tricks. The buzz-hiss decision relies solely on the continuity of the pitch track, because standard indicators are often unreliable in the presence of noise and distortion. The algorithm has been incorporated into a real-time linear prediction vocoder implemented in the Lincoln Digital Voice Terminal, and is being evaluated on speech that has first been processed through a digital telephone simulator. [Work supported by ARPA.]
The LPC error signal—analysis of its variation with the position of the analysis frame
Authors: Rubiner L. R. , Atal B. S. , Sambur M. R.
Published: October 2005

The LPC normalized error provides a measure of the success of linear prediction analysis in modeling a speech signal. Very little is known about the variation of the normalized LPC error as a function of the position of the analysis frame. In this talk we show that the LPC normalized error shows a substantial sample-to-sample variation for voiced speech in all three LPC analysis methods—i.e., the covariance method, the autocorrelation method, and the lattice method. The implication of this result is that standard methods of LPC analysis are often inadequate in that the error signal is uniformly sampled at a low rate (on the order of 100 Hz) leading to aliased results. For applications such as word recognition with frame-to-frame distance calculations using the normalized error [Itakura, IEEE Trans. Acoust. Speech Signal Process, Feb. (1975)], the errors due to uniform sampling can be severe. For speech synthesis applications the effect of uniform sampling of the error signal is a small, but noticeable roughness in the synthetic speech. Various strategies for minimizing the aliasing will be discussed.
Speech-quality testing of variable frame rate (VFR) linear predictive (LPC) vocoders
Authors: Huggins A. W. F. , Viswanathan R. , Makhoul J.
Published: October 2005

VFR transmission of LPC vocoder parameters is a technique developed to reduce the average transmission rate without appreciable loss of quality. The technique transmits parameters at a variable rate in accordance with the changing characteristics of the speech signal. In order to assess the effectiveness of VFR transmission, we performed a pilot experiment to compare it with two other methods for reducing the bit rate: (a) reducing the number of poles, and (b) increasing the quantization step size of the LPC parameters (log area ratios). Thirty-two stimulus sentences were prepared by passing four utterances (two sentences×two speakers) through eight vocoder systems in a 2×2×2 factorial design; two values were assigned to each of the three parameters: average frame rate, number of poles, and quantization step size. Eight listeners made seven-point category ratings of quality degradation. The results of the experiment show that, of the three methods studied, the VFR technique produced the highest quality at any given transmission rate. [Work sponsored by ARPA-IPTO.]
Speech encryption by manipulations of LPC parameters
Authors: Sambur M. R. , Jayant N. S.
Published: October 2005

This paper discusses several manipulations of LPC (linear predictive coding) parameters for providing speech encryption. Specifically, the paper considers temporal rearrangement or scrambling of the LPC code sequence, as well as the alternative of perturbing individual samples in the sequence by means of pseudorandom additive or multiplicative noise. The latter approach is believed to have greater encryption potential than the temporal scrarabling technique, in terms of the time needed to “break the secrecy code.” The encryption techniques were assessed on the basis of perceptual experiments, as well as by means of a quantitative assessment of speech-spectrum distortion, as given by an appropriate “distance” measure.
Effect of room noise and circuit noise on speech-quality ratings over operator-type headsets
Authors: Pierce Linda , Pavel Michael
Published: October 2005

Categorical judgments of communication quality over operator-type headsets were obtained as a function of room noise, circuit noise, and speech volume. The room noise, recorded at an airline reservation center, was similar to wide-band pink noise and consisted primarily of voice babble and teletype-writer noise. It was played back in the test room at levels from 55 to 73 dBA with the spectrum matched to that of the original room noise. The circuit noise, ranging from 37 to 76 dB SPL filtered by a telephone network, extended over a band from 300 to 3300 Hz. The speech was also within this band and ranged in level from 85 to 97 dB SPL. The stimuli used were PB sentences and all speech/circuit noise ratios were at high intelligibility levels. The data indicate that both room noise and circuit noise degrade communication quality ratings. The relative effects of the room noise on the headset ear and the uncovered ear were different for the two headsets compared.
Simplified method of party-wall testing
Authors: Hilliard John K. , Bricken Gordon L.
Published: October 2005

The California Noise Insulation Standards has for the first time imposed STC testing on a broad segment of the building industry. Local officials and builders, its well as consultants, find it increasingly burdensome to implement standard STC (ASTM) field testing because of costs and delays. Inspectors are reluctant to require testing because of the costs and delays, yet there seems to be no easy alternative available. This paper discusses the results of over 100 tests conducted in the Southern California area of party walls with the object of considering whether a screening method for party-wall checks can be used as a viable alternative to the ASTM method in field inspections. The paper will propose a possible technique for adoption as a supplement to ASTM methods currently employed.
The Harpy Speech Recognition System: performance with large vocabularies
Authors: Lowerre B. , Reddy R.
Published: October 2005

The Harpy System [B. Lowerre and R. Reddy, Harpy, a connected speech recognition system, J. Acoust. Soc. Am. 59, S 97 (A) (1976)] has been extended to run with large vocabularies. The system was recently tested with a 1011-word vocabulary language permitting about 1012 possible sentences for an Information Retrieval task using a computerized data base. The system achieved 93.77% word recognition accuracy (89.870 sentence recognition accuracy) on 284 connected speech sentences (containing 1580 words) about 3-sec duration each for a single speaker. The system requires about 12 Mipss (million instructions per second of speech) and uses about 200 000 words of memory on a PDP-10 system. More complete results, including several speakers and additional sentences, will be reported. [Reported supported by the Defence Advanced Research Projects Agency.]
Electronic demonstrations for the teaching of acoustics
Authors: Hartmann W. M.
Published: October 2005

Courses in elementary acoustics taught at many universities provide introductions to the following six topics: physical acoustics, environmental acoustics, the physiology of hearing, auditory psychophysics, musical acoustics, and the nature of human speech. Electronic demonstrations can be used to enhance instruction in each of the six areas. For example, the modules of an electronic music synthesizer together with an audio system and conventional electronic instrumentation form a versatile laboratory for experiments in sound. Among the demonstrations which can be done are the following. Wave interference can be demonstrated by extracting the separate signals from a stereo recording of signals A + B and A − B. Tone and noise bursts can be used to find the speed of sound, echo and reverberation times. Filter banks, set according to mean audiometric presbycusis data, can show the effects of aging on the frequency response of human hearing. Gently nonlinear electronic devices aid the recognition of combination tones normally generated by the inner ear. Analog delay lines can create repetition pitch from a source of white noise. Sine wave signals from many oscillators can be summed and varied to study the perception of pitch of an inharmonic tone complex like that of a chime. The phase variation of a single rectangular wave with pseudorandom pulse width modulation simulates chorus effect. Much of the effect of nonlinear mode mixing in conventional musical instruments can be simulated by patched of linear electronic components, transient generators and voltage controlled signal processing modules. Three resonant filters allow the synthesis of vowel sounds. Frequency shifted or ring modulated speech sounds can exhibit the importance of vowel formant transitions. Intelligibility experiments with chopped speech serve as an introduction to redundancy, autocorrelation and information theory. We note that one does not pursue this informal experimenting with signal processing equipment for long before one discovers effects which are hard to understand. Further pursuit can lead to new results. For example, our intelligibility experiments with chopped speech suggest that redundancy, measured in spoken English text agrees with that found by Shannon for written text.
Project laboratory for a course on the physics of music
Authors: Shonle John I.
Published: October 2005

The spectrum of laboratory styles ranges from a different exercise every week, complete with “cookbook” instructions, to research projects. The advantages of the latter style include high student enthusiasm, a better understanding of the methods of experimentation, and a much deeper knowledge of the chosen area. Our students, usually working in pairs, investigate a single area of acoustics for the semester. Topics are chosen by the students in consultation with the instructor. Examples of topics which have been undertaken in the last four years and method of organization will be given.
Modal density function and number of propagating modes in ducts
Authors: Rice E. J.
Published: October 2005

Often raised questions in duct sound propagation studies involve the total number of propagating modes, the number of propagating radial modes for a particular spinning lobe number, and the number of modes possible between two given values of cutoff ratio or eigenvalue. These questions can be answered approximately by using the modal density function for ducts in a manner similar to that previously published for architectural acoustics. The modal density functions are given for rectangular and circular ducts with a uniform steady flow. Results from this continuous theory are compared to the actual (discrete) modal distributions.
Flight effects on exhaust noise for turbojet and turbofan engines—comparison of experimental data with prediction
Authors: Stone James R.
Published: October 2005

Recent experiments on the effects of flight on jet engine exhaust noise have produced apparently conflicting results. Some of these results do not agree with projections based on classical jet noise theories nor with experimental results from model jet simulated flight tests. It has been shown that in some of the cases reported, the proper corrections were not made to account for the distributed nature of the jet noise sources. It is shown herein that the remaining discrepancies can be reconciled by considering the combined effects of jet-mixing noise, internally generated engine exhaust noise, and shock noise. This paper demonstrates that static and in-flight jet engine exhaust noise can be predicted with reasonable accuracy when the multiple-source nature of the problem is taken into account. Jet mixing noise and shock noise are predicted from an improved version of the NASA interim prediction method. A provisional method of estimating internally generated noise is used, based partly on existing prediction methods and partly on recently reported engine data.
Effects of forward velocity on noise for a J85 turbojet engine with multi-tube suppressor from wind tunnel and flight tests
Authors: Stone J. R. , Miles J. H. , Sargent N. B.
Published: October 2005

In order to obtain an understanding of the effects of forward velocity on exhaust noise suppression, flight and wind-tunnel noise tests were conducted using a J85 turbojet engine. Nozzle configurations of primary interest were a 104-tube suppressor with and without an acoustically treated shroud. Engine installation was as nearly idential as possible for both flight and wind tunnel tests. Some differences necessarily existed, however; e.g., for the flight tests the engine was in motion with respect to the microphones, and for the wind-tunnel tests it was not. Exact simultaneous matching of engine speed, exhaust velocity, and exhaust temperature was not possible, and the wind-tunnel maximum Mach number was ∼0.27, while the flight Mach number was ∼0.37. The nominal jet velocity range was 450–640 m/sec. For both experiments, background noise limited the jet velocity range for which significant data could be obtained. The observed forward velocity effects for the suppressor indicate that jet noise has been suppressed to a level where the in-flight directivity is similar to that expected for internally generated noise.
Simulation of flight-type engine fan noise in the NASA—Lewis 9×15 anechoic wind tunnel
Authors: Heidmann M. F. , Dietrich D. A.
Published: October 2005

A major problem in the measurement of aircraft engine fan noise is the difficulty in simulating in a ground-based facility the noise that occurs during flight. Flight-type noise as contrasted to the usual ground-static noise exhibits substantial reductions in (1) the time unsteadiness of tone noise and (2) the mean level of tones calculated to be nonpropagating or cutoff. A model fan designed to have a nonpropagating tone was acoustically tested in the anechoic wind tunnel under both static and tunnel flow conditions. The properties that characterize flight-type noise were progressively simulated with increasing tunnel flow. The distinctive lobed directivity patterns of propagating duct modes were also observed. The acoustical data with tunnel flow appear uncontaminated by the excess noise attributed to the ingestion of flow disturbances that prevail near most static test facilities. This anechoic wind tunnel appears to be a useful facility for flight applied acoustic research on aircraft engine fans.
Macroscopic study of time unsteady noise of an aircraft engine during static tests
Authors: Clark B. J. , Heidmann M. F. , Kreim W. J.
Published: October 2005

Static tests of aircraft engines can exhibit greater than 10-dB random unsteadiness of tone noise levels because flow disturbances that prevail near test site facilities are ingested. Presumably such changes are related to installation and test site features. This paper presents some properties of unsteady noise observed at a NASA—Lewis facility during tests of a Lycoming YF-102 turbofan engine. Time and spacial variations in tone noise obtained from closely spaced farfield and inlet duct microphones are displayed. Long (0.5-sec) to extremely short (0.001-sec) intermittent tone bursts are observed. Unsteadiness of the tone, its harmonies, and the broadband noise show little similarity. In the farfield, identity of tone bursts is retained over a directivity angle of less than 10°. In the inlet duct, tone bursts appear to propagate axially but exhibit little circumferential similarity. They show only slight relationship to tone bursts observed in the farfield. The results imply an intermittent generation of random mixtures of propagating duct modes.
Study of the behavior of radiational inversions with a calibrated acoustic sounder
Authors: Kerman B. R. , Turner H. E. , Mickle R. E. , Clink W.
Published: October 2005

A calibrated acoustic sounder developed by the Atmospheric Environment Service of Canada is compared to an in situ sensor measuring the acoustic backscatter cross-section directly, Some deficiencies in the calibration technique and discrepancies with the in situ sensor are analyzed. The vertical structure of the acoustic backscatter cross-section during various stages of a radiational inversion is examined in conjunction with supporting temperature, humidity and velocity data taken from a tethered balloon system.
Upper atmosphere circulations as observed with the use of natural infrasound
Authors: Rind D.
Published: October 2005

Natural infrasound of 5-sec periods is generated continuously by ocean waves to the east of our location. The amplitude characteristics and horizontal phase velocity are functions of the acoustic reflection characteristics of the upper atmosphere. The monthly patterns of the observed acoustic parameters, culled from eight years of continuous observations, are reviewed and compared to the existing knowledge of 30–120-km winds and temperatures. Included in the discussion are (1) the percentages of observed semidiurnal and diurnal amplitude oscillations for each month, related to the reliability of these tides in the upper atmosphere; (2) annually repeating oddities in each month such as a late March increase in semidiurnal tidal strength and varying meridional wind influences from summer to winter; and (3) the failure of current atmospheric models to correctfy predict the observed lower thermosphere sound velocity at 0400 which is much greater from the east then predicted.
Sound reflection from locally reacting surfaces of finite extent
Authors: Hadden W. James , Vidimos Robin A. , Sencil Philip M.
Published: October 2005

An approximation for the scattering of spherical waves by a rectangular patch with finite acoustic impedance is presented. Comparisons are made with laboratory experiments in which sound pressure levels were measured in the direction of specular reflection. Particular attention is given to the way in which both the theoretical and experimental results approach the limiting case of a plane surface with finite acoustic impedance.
Radiation in a random medium
Authors: Wenzel Alan R.
Published: October 2005

A theoretical investigation of the wave field radiated by a point source in a random medium is described. The analysis, which is based on the smoothing method, deals specifically with time-harmonic waves in a medium which is statistically homogeneous and isotropic. Approximate results are obtained for the special case in which the wavelength is much less than the correlation length of the medium. These results show that, in a mean-square sense, the randomness of the medium leads to a slight amplification of the wave field. [Work supported by NASA.]
Mechanical response of the lung at high frequency
Authors: Fredberg J. J. , Hoening A.
Published: October 2005

For the purposes of learning about pulmonary structure, interpretation of high frequency impedance of branching airway systems may be facilitated by a theoretical prediction of input impedance. We have developed ways of treating special classes of asymmetry in branching structures which simulate a variety of observations unaccounted for in current symmetric models. This self-consistent analysis, which assumes that small sub-units can be joined in an appropriate manner to simulate ever-larger branching structures, also incorporates the geometric and dynamic properties of the airway walls and enclosed gas. This analysis has been successfully applied to models of adult and infant lungs, and mirrors trends found in existing data. The model also suggests quantitative and qualitative dependences of system impedance responses to certain airway system characteristics, such as wall properties, branching asymmetry, and central and peripheral airway obstructions. Modal overlap and clumping are observed. Wall properties and branching asymmetry are both found to be important at low frequencies (≲1 kHz), while the asymmetric nature of the structure predominates at higher frequencies (≳3 kHz). The effect of central airway obstructions is most marked at higher frequencies (>2 kHz) where the number of apparent modes has decreased as modal regularity increases. Peripheral airway obstructions, in contrast, have small effect at high frequency.
Acoustic nonspecular scattering from a plate with line impedance discontinuities
Authors: Stephanishen Peter R.
Published: October 2005

An approach is presented to investigate the acoustic nonspecular bistatic scattering of an incident acoustic plane wave by a fluid loaded plate with line impedance discontinuities. The approach is based on the extension of a method which was presented in an earlier paper [P. R. Stepanishen, J. Acoust. Soc. Am. 59, S32(A) (1976)] to investigate the effect of a single discontinuity on the acoustic transmission and scattering from a plate. The total scattered field is expressed as a sum of a specular reflected plane wave and a sum of scattered cylindrical waves which originate at the discontinuities on the plate which is described by Timoshenko-Mindlin plate theory. The scattering strengths of the cylindrical waves can exhibit highly directional characteristics and are expressed, using Fourier transform methods, as the solution of a linear set of coupled equations. Several special cases are investigated and discussed. In particular, it is noted that when the effects of the acoustic-elastic interactions between the scatterers or impedance discontinuities can be neglected, the nonspecular scattered field can be expressed as a product of two terms, i.e., the nonspecular scattered pressure from a single discontinuity times an equivalent scattering array factor.
Effects of forward masking on detection and intensity discrimination of pulsed sinusoids
Authors: Wildin Gregory P. , Viemeister Neal F.
Published: October 2005

The effect of the intensity of a pure-tone masker on the forward-masked threshold of a brief pure-tone signal of the same frequency was investigated using masker level up to 100 dB SPL. Results agree with and extend those of Houtgast [in Facts and Models in Hearing, edited by E. Zwicker and E. Terhardt (Springer, Berlin, 1974), pp. 258–265 and are described by a square-root relation between masker and signal level, to masker levels of at least 80 dB. In contrast, Weber's law is approximately valid for intensity discrimination of pulsed sinusoids in the presence of a pure-tone forward masker when the ratio of masker level of the standard is fixed. For fixed intensities above the detection threshold in forward masking, the forward-masked increment threshold generally increases only slightly with increases in masker level. [Research supported by the Center for Research in Human Learning and by NSF.]
Suppression in pulsation-threshold patterns
Authors: Shannon Robert V.
Published: October 2005

In a pulsation, threshold experiment the frequencies and intensities of the two tones in the “masker” were fixed and the frequency of the signal was varied. The plot of pulsation threshold versus signal frequency is called the pulsation-threshold pattern of the “masker”. First the pulsation-threshold pattern was measured when the “masker” contained only a 1000-Hz tone. Then the patterns were measured for a 400-Hz tone and for a 1500-Hz tone. Finally, the pattern was measured when the masker contained both the 1000-Hz tone and either the 400-Hz tone or the 1500-Hz tone. The 400-Hz tone was able to reduce the pulsation thresholds around 1000-Hz by 20 dB or more for each of the three subjects. The 1500-Hz tone did not reduce the pulsation thresholds at 1000 Hz but did at 1100 Hz, i.e., on the high-frequency tail of the 1000-Hz pattern. Suppression is assumed to be the cause of these reductions. Thus one tone is able to suppress only part of the excitation pattern of another tone. Mutual suppression was also indicated for one subject. Relationships to partial masking will be discussed. [Work supported the NINCDS.]
Pitch strength of ripple noise
Authors: Yost William A. , Hill Richard
Published: October 2005

Ripple noise is produced when a broadband noise is delayed (T) and either added to or subtracted from the original noise. The resulting waveform has a power spectrum in which power varies cosinusoidally as a function of frequency. Ripple noise has been used to study pitch perception and echo processing. In the present experiments subjects were asked to discriminate between two noises which differed in the amount of delay introduced (one stimulus continued a delay T the second stimulus a delay T + 0.09T). The discriminability of these two noises was studied as a function of the amount of attenuation added to the delayed noise. Discriminations between these two noises could be made when T ranged from 0.5 to 20 msec before the subjects could no longer make the discrimin-delayed noise when the delays were in the range from 2 to 5 msec before the subjects could no longer make the discriminations. More attenuation could be added to the delayed noise for threshold discrimination when it was added back to the original source than when it was subtracted from the original source than when it was subtracted from the original source. Discrimination did not change as a function of octave band filtering the ripple noise when the center frequency of the filters varied from 200 to 3200 Hz. The results will be discussed in terms of pitch perception, especially in regard to the concept of a spectral dominance region for pitch and pitch strength. The data will also be related to the echo processing abilities of the auditory system. [Work supported by NSF.]
Dichotic lag effect and right-ear advantage in children 7 to 15
Authors: Mirabile P. J. , Porter R. J. , Hughes L. F. , Berlin C. I.
Published: October 2005

Children aged 7, 9, 11, 13, and 15 years were tested on their ability to identify simultaneous and time-staggered dichotic consonant-vowel (CV) stimuli. At each age level, 15 males and 15 females listened to 300 dichotic trials, 60 trials at each of five onset asynchronies: 0, 15, 30, 60, and 90 msec. Simultaneous-onset results confirmed earlier dichotic findings in children: age-related performance increases and superior right-ear performance. In addition, the data revealed a lag effect with temporally asynchronous stimuli in a pattern similar to adults. However, the lag effect in younger children was more pronounced at 90 msec than at the 30 msec usually seen for adults. [Supported by NIH.]
Dichotic and monotic masking of CV's by second-formants with different transition starting values
Authors: Whitaker R. G. , Porter R. J.
Published: October 2005

Subjects identified synthetic, two-formant, voiced stop-vowel “targets” selected from an 11-step [bæ]—[gæ] continuum. Three targets were selected from the center of the “b”, “d” and “g” categories, the two others from near category boundaries. Targets (presented at 50 dB) were simultaneously paired with eleven second-formants (“bleats”) extracted from the CV continuum stimuli. Pairs of targets and bleats were presented, dichotically and monotically in blocks, in a within-subjects design. Overall intensity of the bleat relative to the target was varied over blocks in 6-dB steps from 0 to −24 dB. Subjects' performance varied with the nearness of the targets to boundary and the position of the bleat along the continuum; greatest interference occurred with the highest bleat intensities. Dichotic results tended to show a greater sensitivity to the “phonetic relevance” of the bleats and targets as defined in terms of their places on the continuum whereas monotic results tended to reflect more factors such as “energy averaging” in the F2 region. [Supported by NIH.]
Distinctive feature analysis of binaural versus monaural consonant perception as a function of signal to noise ratio
Authors: Yonovitz A. , Thompson C. L. , Dickenson Priscilla C. , Lozar J. T.
Published: October 2005

Electret microphones inserted in the ear canals were used to binaurally record noncorrelated noise from two speakers equidistant at ±45° azimuth in an anechoic chamber. Twenty-one CV's (with “a) were similarly recorded at 0° azimuth. Various signal to noise ratios were obtained by mixing each binaural recording. These recordings maintained the effects of head shadow, pinna, etc. At each signal to noise ratio subjects wrote their response to the 30 presentations of each CV binaurally and monaurally. Confusion matrices were constructed. Data were analyzed in terms of consonant confusions and feature intelligibility. Multivariate analysis indicated differences in the contribution of the various features when binaural and monaural data were considered and also revealed changes in this pattern as a function of signal to noise ratio.
Connected Digit Recognition using symbolic representation of pronunciation variability
Authors: Goodman G. , Lowerre B. , Reddy R. , Scelza D.
Published: October 2005

Most connected speech recognition systems such as Harpy and Hearsay-II use some form of symbolic representation alternative pronunciations of the vocabulary whereas most isolated word recognition systems use word templates. In an attempt to compare relative performance of systems that use symbolic representations of words, the Harpy system was run on a connected digit task requiring the recognition of random three-digit sequences. Each of ten speakers (seven male and three female) spoke 30 training sentences and 100 test sentences over a period of two weeks in a computer terminal room environment (approximately 65 dBA). Using speaker-dependent phoneme templates, the word error rate over all the ten speakers was about 2%. Using speaker-independent phoneme templates computed from the training data for all the speakers (male and female), the word error rate was about 8% for a test data set of 1200 random connected three-digit sequences from 20 speakers (including ten new speakers). The recognition time is about 4.5 Mipss (million instructions per second of speech). [Research supported by the Defense Advanced Research Projects Agency.]
Parametric representation of speech
Authors: Gill G. , Reddy R.
Published: October 2005

As digital processing of speech becomes commonplace, it becomes desirable to have a parametric representation of speech which is simple, fast, accurate, and directly obtainable from the PCM representation of speech. The ZAPDASH representation of speech (Zerocrossings And Peaks of Differenced And Smooth waveforms) is one such. The PCM data is used to generate a different waveform and a down sampled, smoothed waveform (for 10-kHz sampling rate, the smoothing FIR filter coefficients were − 1 0 1 2 4 4 4 2 1 0 − 1, used every fourth point). Peak-to-peak distances and number of zerocrossings are calculated each 10 msec, resulting in 400 8-bit parameters per second of speech. ZAPDASH can be done in 15–20 computer instructions per sample and can be extracted in less than a 1/3 real time on minicomputers with 2 μsec instruction time. Although this representation is not noticeably different other similar proposals, it seems to be fairly robust and accurate, and is used in the feature extraction, segmentation, and labeling parts of the Harpy and Hearsay-II systems. Fortran and PDP-11 machine language versions are available from the authors. [Research supported by the Defense Advanced Research Projects Agency.]
The HWIM speech understanding system—Overview and performance
Authors: Wolf Jared J. , Woods William A.
Published: October 2005

HWIM (for Hear What I Mean), the speech understanding system developed at BBN as part of the recent five-year ARPA Speech Understanding Research Project, is designed to “understand” naturally spoken utterances relevant to a task domain of travel budget management. Its vocabulary is over 1000 words, and its grammar permits a habitable subset of natural English. HWIM contains sources of knowledge at the levels of acoustic-phonetics, phonology, vocabulary, syntax, semantics, factual knowledge, and discourse. This paper describes the system as a whole and presents its performance results at the end of the ARPA project. [This research was supported by the Advanced Research Projects Agency of the Department of Defense and was monitored by ONR under Contract No. N00014-75-C-0053.]
Phonetic and lexical processing in the HWIM speech understanding system
Authors: Schwartz Richard M. , Klovstad John W. , Zue Victor W. , Makhoul John I. , Wolf Jared J.
Published: October 2005

The “front end” of HWIM, the BBN speech understanding system, is that part of the system that governs the formation and evaluation of hypotheses between the levels of the speech signal and the word. It comprises processes for signal processing, acoustic-phonetic recognition, lexical-segmental matching, and lexical-parametric matching. Implicit in the lexical matching processes is the application of phonological rules, both within word pronunciations and across word boundaries. A consistent scoring policy governs the evaluation of hypothesis at the segmental and word levels, and this policy is carried into the control component of the system, where it is applied to multiword hypotheses about the interpretation of the utterance. [This research was supported by the Advanced Research Projects Agency of the Department of Defense and was monitored by ONR under Contract No. N00014-75-C-0053.]
Comparative evaluation of the AX and two ABX procedures
Authors: Pastore R. E. , Friedman C. J. , Baffuto K. J.
Published: October 2005

The ABX procedure, which has been used extensively in speech perception research, provides a great deal of face validity. Unlike the classic same—different (or AX) procedure, each member of the pair of stimuli to be discriminated are presented on every trial. Furthermore, in theory, the estimates of discriminability are relatively free from response bias. Recently, this procedure has received considerable criticism for being contaminated, on a theoretical basis, by memory effects and thus providing diminished estimates of the discriminability between stimuli. In a series of experiments, we compared (a) the typically employed “random” ABX procedure (random relative relationship between A and B), (b) a “fixed” ABX procedure (fixed relative relationship between A and B), and (c) the same—different (AX) procedure in terms of psychometric functions and isosensitivity contours using an intensity discrimination task. The results are discussed in terms of the relative sensitivity of the procedures and their susceptibility to contamination by response bias.
An interdepartmental musical acoustics course
Authors: Hall Donald E. , Hatton Gaylen
Published: October 2005

We believe it is important not to slight either music or physics in teaching a musical acoustics course. We are currently offering a course with two lectures (by G. H., with some informal participation by D. H.) and one coordinated lab/activity session (by D. H.) each week. We will discuss course organization, activity topics, advantages, and disadvantages of this plan. As an important component of the activities, each student works on an independent project and presents a report to the class. We conclude (1) that inclusion of experiments and activities as well as demonstrations is highly preferable to lectures alone, and (2) the course is more valuable and interesting when the teacher(s) can provide real expertise in both music and physics.
Musical performance modification in differing acoustic environments
Authors: Harrison D. B.
Published: October 2005

By modifying their techniques, musicians are able to adapt performances to a variety of acoustic environments. Under some circumstances a musician is able to convey to the listener a property of the sound field which is nonexistent. Specific techniques practiced by organists are discussed and illustrated with recorded examples. The logic of many of these practices developed in the past may be seen in the light of modern acoustic and psychoacoustic knowledge. (It is suggested that as a supplement to courses in musical acoustics, material oriented more towards the performer's viewpoint, material such as this, might entice many musicians to learn more of the science of their art.)
Peruvian whistling bottles
Authors: Garrett Steven , Star Daniel K.
Published: October 2005

Measurements were made of the frequency and intensity of 73 pottery “whistling bottles” encompassing a 200-year period from 500 B. C. to 1550 A. D. representing nine pre-Columbian cultures. We have found the whistles group acoustically by culture. The bottles are currently regarded by anthropologists as utilitarian liquid containers with the whistle providing an amusing method of venting to facilitate the passage of air during the pouring and filling of liquids. We are suggesting an alternative interpretation of the bottles as ritual artifacts specifically produced as whistles. We base this interpretation on the close physical examination of a large sample and particularly on the pronounced effect of low frequency difference tones. If two or more whistles from a single culture are played simultaneously, difference tones are created within the ear due to the nonlinear interaction of their high-intensity tones.
Trumpet acoustics: Correcting intonation faults
Authors: Cardwell W. T.
Published: October 2005

Calculations are possible of the locations and sizes of constrictions and enlargements that may be put into a trumpet air column for correcting intonation errors of the trumpet. In certain respects, and to at least a useful approximation, the trumpet behaves as a simple, closed-open, cylindrical resonator, and perturbation theory applied to a closed-open cylinder model permits the calculations of the locations and sizes of the constrictions and enlargements. The corrector-amplitude versus distance function is a Fourier transform of the experimentally determined intonation-error versus mode function. An interesting experiment, making use of the theory, is the raising of the (usually) flat fifth mode without significantly affecting any of the other playing modes.