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Showing papers in "Journal of the Acoustical Society of America in 1969"


Journal ArticleDOI
TL;DR: In this article, the authors introduced the outgoing spherical (or circular cylinder) partial waves as a basis for the equation QT = − Re (Q) describing scattering for general incidence on a smooth object of arbitrary shape.
Abstract: Upon introducing the outgoing spherical (or circular cylinder) partial waves {ψn} as a basis, the equation QT = − Re (Q) is obtained for the transition matrix T describing scattering for general incidence on a smooth object of arbitrary shape. Elements of Q involve integrals over the object surface, e.g. Qmn = ±(i2)δmn+(k8π)∫dσ⋅∇[Re(ψm)ψn]. where the −, + apply for Dirichlet and Neumann conditions, respectively. For quadric (separable) surfaces, Q is symmetric. Symmetry and unitarity lead to a secular equation defining eigenfunctions for general bodies. Some apparently new closed‐form results are obtained in the low frequency limit, and the transition matrix is computed numerically for the infinite strip.

469 citations


Journal ArticleDOI
TL;DR: An attempt is made to develop a quantitative theory of intensity resolution that is applicable to a wide variety of experiments on discrimination, identification, and scaling and provides, among other things, a new interpretation of the 7 ± 2 phenomenon.
Abstract: An attempt is made to develop a quantitative theory of intensity resolution that is applicable to a wide variety of experiments on discrimination, identification, and scaling. The theory is composed of a Thurstonian decision model, which separates sensitivity from response bias, and an internal‐noise model, which separates sensory limitations from memory limitations. It is assumed that the subject has two memory operating modes, a sensory‐trace mode and a context‐coding mode, and that the use of these two modes is determined by the characteristics of the experiment. In one‐interval paradigms, it is assumed that the context‐coding mode is used, and the theory relates resolution to the total range of intensities in the stimulus set. In two‐interval paradigms, it is assumed that the two modes are combined, and the theory relates resolution to both the total intensity range and the duration between the two intervals. The theory provides, among other things, a new interpretation of the 7 ± 2 phenomenon.

440 citations


Journal ArticleDOI
TL;DR: In this article, a male speaker recorded monosyllabic words and a continuous sentence and a pitch-synchronous analysis was carried out by a digital computer on the vowel portions of these samples, for every pitch period, the analysis provided: formant frequencies, waveform of the glottal excitation function, and an accurate pitch-period measurement.
Abstract: An experiment is described that investigates listener preferences for speech samples with varying glottal pulse‐shape parameters. A male speaker recorded monosyllabic words and a continuous sentence. A pitch‐synchronous analysis was carried out by a digital computer on the vowel portions of these samples. For every pitch period, the analysis provided: formant frequencies, waveform of the glottal excitation function, and an accurate pitch‐period measurement. In each vowel, the natural glottal excitation function was replaced by a mathematical function with a shape not unlike that of natural glottal waves. The waveform of the artificial function could be modified by varying two parameters analogous to the “opening” and “closing” times of natural glottal pulses. Four opening times and four closing times (fixed relative to individual pitch‐period lengths) were selected for experimentation. The artificial functions were substituted for the natural glottal pulses, and the speech wave was reconstituted period by period. Subjective evaluation of the reconstituted speech was carried out by means of computer‐controlled paired‐comparison tests. Sequential testing strategies were used to reveal the synthetic samples perceptually most similar to the original. The results indicate that an artificial glottal wave having the same analytical specification in every pitch period can yield reconstituted speech of quality comparable to natural speech.

363 citations


Journal ArticleDOI
TL;DR: Five descriptive parameters of hearing—high‐frequency and low‐frequency sensitivity, lowest threshold, best frequency, and area of the audible field—are compared statistically, first, among mammals in general, and then, among seven animals selected to approximate a phylogenetic sequence of man's ancestors.
Abstract: Five descriptive parameters of hearing—high‐frequency and low‐frequency sensitivity, lowest threshold. best frequency, and area of the audible field—are compared statistically, first, among mammals in general, and then, among seven animals selected to approximate a phylogenetic sequence of man's ancestors. Three potentially explanatory parameters body size, maximum binaural time disparity, and recency of common ancestry with man—are also explicitly included in the analysis. The results show that: high‐frequency hearing (above 32 kHz) is a characteristic unique to mammals, and, among members of this class, one which is commonplace and primitive. Being highly correlated with functionally close‐set ears, it is probably the result of selective pressure for accurate sound localization. Low‐frequency hearing improved markedly in mankind's line of descent, but the kind and degree of improvement are not unique among mammalian lineages. High sensitivity developed in the earliest stages of man's lineage and has remained relatively unchanged since the simian level. The frequency of the lowest threshold has declined in Man's lineage—the greatest drop probably occurring during the Eocene. The total area of the audible field increased until the Eocene and has decreased since then.

310 citations


Journal ArticleDOI
TL;DR: In this article, the acoustic radiation force on a solid sphere suspended freely in a plane progressive sound field is calculated taking into account the elasticity of the sphere, and the accuracy of absolute acoustic intensity determination by the radiation force method is improved by the use of the present theory instead of King's.
Abstract: The acoustic radiation force on a solid sphere suspended freely in a plane progressive sound field is calculated taking into account the elasticity of the sphere. The radiation force versus ka curve, ka being the radius times the wavenumber in the surrounding medium, shows sharp maximums or minimums corresponding to resonance in every normal mode of free vibration of the sphere. The measurements made in our laboratory are in good agreement with the theory. The accuracy of absolute acoustic‐intensity determination by the radiation‐force method will be improved by the use of the present theory instead of King's.

301 citations


Journal ArticleDOI
TL;DR: In this paper, a data reduction procedure based on the Karhunen-Loeve representation was used to represent the pitch information in each contour in a 20-dimensional space.
Abstract: The results of a study aimed at finding the importance of pitch for automatic speaker recognition are presented. Pitch contours were obtained for 60 utterances, each approximately 2‐sec in duration, of 10 female speakers. A data‐reduction procedure based on the Karhunen‐Loeve representation was found effective in representing the pitch information in each contour in a 20‐dimensional space. The data were divided into two portions; one part was used to design the speaker recognition system, while the other part was used to test the effectiveness of the design. The 20‐dimensional vectors representing the pitch contours of the design set were linearly transformed so that the ratio of interspeaker to intraspeaker variance in the transformed space was maximum. A reference utterance was formed for each speaker by averaging the transformed vectors of that speaker. The test utterance was assigned to the speaker corresponding to the reference utterance with the smallest Euclidean distance in the transformed space. The percentage of correct identifications using the above procedure was found to be 97%. The recognition rate increased to 98% with the addition of duration as an independent measure.

297 citations


Journal ArticleDOI
TL;DR: A computational algorithm for estimating pitch periods of speech in the time domain is presented, and two recent modifications of the algorithm are discussed in detail.
Abstract: A computational algorithm for estimating pitch periods of speech in the time domain is presented, and two recent modifications of the algorithm are discussed in detail. The algorithm and its modifications have been found to be relatively accurate and efficient in tests on real and synthetic speech.

283 citations


Journal ArticleDOI
TL;DR: In this paper, the authors investigated the problem of obtaining an analytic solution and practical computational procedures for recovering the properties of an unknown elastic medium from waves that have been reflected by or transmitted through the medium.
Abstract: This paper investigates the problem of obtaining an analytic solution and practical computational procedures for recovering the properties of an unknown elastic medium from waves that have been reflected by or transmitted through the medium. The medium consists of two homogeneous half‐spaces in contact with a heterogeneous region. The analytic solution is obtained by transforming the equation of motion for the propagation of plane waves at normal incidence in a stratified elastic medium into a one‐dimensional Schrodinger equation for which the inverse‐scattering problem has already been solved. The practical computational procedures are obtained by solving the corresponding discrete inverse‐scattering problem resulting from approximating the heterogeneous region with a sequence of homogeneous layers such that the travel time through each layer is the same. In both the continuous and discrete inverse scattering problems, the impedance of the medium as a function of travel time is recovered from the impulse response of the medium. A discrete analogy of the continuous solution is also developed. Similar results are obtained for a stratified elastic half space bounded by a free surface.

252 citations


Journal ArticleDOI
TL;DR: A quantitative psychophysiological theory is developed for loudness level and loudness as a function of stimulus duration that shows how this can be achieved in spite of a nonlinear relationship between sound intensity and neural excitation.
Abstract: A quantitative psychophysiological theory is developed for loudness level and loudness as a function of stimulus duration. It is based on the psychophysical as well as neurophysiological evidence that the apparent temporal summation of acoustic energy is a result of neural summation at a high level of the auditory system. The theory shows how this can be achieved in spite of a nonlinear relationship between sound intensity and neural excitation. The temporal decay of neural firing preceding the final stage of temporal summation seems to be responsible for overcoming the nonlinearity.

249 citations


Journal ArticleDOI
TL;DR: Results from three experiments that employed various permutations of the aforementioned conditions are reported, and it is shown that mixing one speech train with noise (either modulated or unmodulated) induced about 3.2 dB excess masking.
Abstract: Shifts in masked spondee thresholds during several conditions of listening (monaural, homophasic, antiphasic, and with interaural time disparity) in the presence of one to four competing maskers were measured. The maskers used were white noise, white noise modulated four times per second by 10 dB with a 50% duty cycle, the same noise with 75% duty cycle, connected speech by one male talker, and connected speech by a second male talker. Results from three experiments that employed various permutations of the aforementioned conditions are reported. The findings, after equating conditions to equivalent masker levels, were four. First, the modulated noise with 50% duty cycle produced about 3.5 dB less masking than that produced by unmodulated white noise. Second, the modulated noise with 75% duty cycle allowed only about 1 dB less shift than did the unmodulated noise. Third, mixing one speech train with noise (either modulated or unmodulated) induced about 3.2 dB excess masking. This excess is here termed per...

232 citations


Journal ArticleDOI
TL;DR: This consonance characteristic turned out to show a simple V curve if consonance is plotted against the frequency percent deviation in a logarithmic scale, and it suggests a dynamic and a static factor in consonance perception.
Abstract: Extensive psychological experiments were carried out in this Part I on the consonance sensation of various dyad tones consisting of two components. As the frequencies of two components f1 and f2 (with an equal SPL) separate, the consonance gradually decreases down to the most dissonant point, whereafter it monotonically increases and mostly recovers at an octave separation. This consonance characteristic turned out to show a simple V curve if consonance is plotted against the frequency percent deviation in a logarithmic scale, and it suggests a dynamic and a static factor in consonance perception. The most dissonant frequency deviation is approximately 10% at f1=440 Hz, and it increased with sound pressure and frequency, but it was not simply proportional to the critical bandwidth. For dyads whose two components have different sound‐pressure levels (SPLs) L1 and L2, the consonance differs with spectrum forms, even when the level difference is the same. A dyad with a spectrum form (P1>P2, f1

Journal ArticleDOI
TL;DR: Experiments were carried out to investigate the correlation between the perceptual and physical space of 11 vowel sounds and yielded an excellent result with correlation coefficients of 0.992, 0.971, and 0.742 along the corresponding dimensions.
Abstract: Experiments were carried out to investigate the correlation between the perceptual and physical space of 11 vowel sounds. The signals were single periods out of the constant vowel part of normally spoken words of the type h (vowel) t, generated continuously by computer. Pitch, loudness, onset, and duration were equalized. These signals were presented to 15 subjects in a triadic‐comparison procedure, resulting in a cumulative similarity matrix. Multidimensional scaling (Kruskal) of this matrix resulted in a three‐dimensional perceptual space with 1.6% stress. The signals were also analyzed physically with 13‐oct band filters. Principal‐components analysis of the decibel values per frequency band indicated that three dimensions accounted for 81.7% of the total variance. Matching the perceptual and the physical configurations to maximal congruence yielded an excellent result with correlation coefficients of 0.992, 0.971, and 0.742 along the corresponding dimensions. The formant frequencies and levels were co...

Journal ArticleDOI
TL;DR: In this article, the Cramer-Rao technique is used to set a lower bound on the rms bearing error and the results are compared with the bearing error of a slightly modified split-beam tracker.
Abstract: This paper studies the minimum bearing error attainable with a linear passive array. Signal and noise are stationary Gaussian processes with arbitrary power spectra, and the noise is assumed to be statistically independent from hydrophone to hydrophone. The Cramer‐Rao technique is used to set a lower bound on the rms bearing error and the results are compared with the bearing error of a slightly modified split‐beam tracker. The latter reaches the lower bound for a two‐element array and comes very close to reaching it for a linear array with an arbitrary number of equally spaced hydrophones. Thus, the modified split‐beam tracker is very nearly optimal for the uniformly spaced array. Comparisons of split‐beam tracker error with the Cramer‐Rao lower bound are also obtained for nonuniform hydrophone spacings.

Journal ArticleDOI
TL;DR: Comparison between the performance of listeners with normal audiograms and those with high‐frequency hearing loss shows this interaction between frequency and the time constant to be similar for both samples.
Abstract: Levels of monaural signals at behavioral threshold were determined by a psychophysical method of adjustment for seven highly trained listeners. Thresholds were studied as a function of signal frequency (octave steps, from 0.125 to 8 kHz) and of signal duration (logarithmic steps, from 16 to 1024 msec). Measurements were made in the presence of a contralateral broad‐band masking noise with a spectrum level of 30 dB SPL. The time constant, τ estimated from at least 12 replications of each measurement, was found to range systematically from values considered normal (125–175 msec) by some earlier investigators, at low frequencies, to much lower values (30–70 msec) at high frequencies. Comparison between the performance of listeners with normal audiograms and those with high‐frequency hearing loss shows this interaction between frequency and the time constant to be similar for both samples. The data are also compared to the results of a second experiment that employed a two‐alternative forced‐choice psychophysical method.

Journal ArticleDOI
TL;DR: The model introduces a new concept of “dissonance intensity” in a certain process of dissonance perception and extends the “power law” to the dissonance sensation, which is not clearly related to a certain physical value.
Abstract: A theory for calculating subjective dissonance of static complex tones has been established. The theory proposes a dissonance perception model that assumes that the mutual interactions between two components constitute an essential additive unit contributing to the dissonance. The model introduces a new concept of “dissonance intensity” in a certain process of dissonance perception and extends the “power law” to the dissonance sensation, which is not clearly related to a certain physical value. Practical calculation procedures are described according to the experimental results of dyads in Part I. Theoretical calculation for various kinds of complex tones showed good agreements with psychological experiments. An application to chords of synthesized harmonic complex tones predicted great dependence of consonance characteristics on the harmonic structures, which are not taken into account in the conventional theory of harmony. It became clear that the fifth (2:3) is not always a consonant interval. A chord ...

Journal ArticleDOI
TL;DR: A digital speech analysis‐synthesis system based on a recently proposed approach to the deconvolution of speech is presented and either a zero‐phase or minimum‐phase characteristic can be obtained by simple weighting of the cepstrum before transformation.
Abstract: A digital speech analysis‐synthesis system based on a recently proposed approach to the deconvolution of speech is presented. The analyzer is based on a computation of the cepstrum considered as the inverse Fourier transform of the log magnitude of the Fourier transform. The transmitted parameters represent pitch and voiced unvoiced information and the low‐time portion of the cepstrum representing an approximation to the cepstrum of the vocal‐tract impulse response. In the synthesis, the low‐time cepstral information is transformed to an impulse response function, which is then convolved with a train of impulses during voiced portions or a noise waveform during unvoiced portions to reconstruct the speech. Since no phase information is retained in the analysis, phase must be regenerated during synthesis. Either a zero‐phase or minimum‐phase characteristic can be obtained by simple weighting of the cepstrum before transformation.

Journal ArticleDOI
TL;DR: In this paper, a coupled mode involving terms decaying rapidly beneath the free surface and a term representing a bulk wave radiating into the solid is introduced, which has many of the properties of a normal surface wave but has a phase velocity higher than that of the transverse bulk wave in the corresponding direction.
Abstract: When the free surface is anisotropic, mode of elastic surface‐wave propagation can arise that has many of the properties of a normal surface wave but has a phase velocity higher than that of the transverse bulk waves in the corresponding direction. The pseudo surface wave is a coupled mode involving terms decaying rapidly beneath the free surface and a term representing a bulk wave radiating into the solid. For many choices of crystal and plane of propagation, the contribution of the bulk term over a range of directions is small enough that the energy of the wave is essentially concentrated within a few wavelengths of the free surface and flows parallel to the surface as with the normal elastic surface waves. Moreover, in certain specific directions, the bulk term disappears completely and the pseudo‐surface wave has all the properties of a normal surface wave. The method of computation of the characteristics of the pseudo surface waves is outlined here and typical results of velocity, displacements and e...

Journal ArticleDOI
TL;DR: In this article, the average and variance of power injected by point sources are calculated and the statistics of response near and away from the driving point are also found, and two kinds of estimation intervals are derived and applied to some simple examples.
Abstract: The calculation of input power and response of simple models of rooms and structures is described. The approach is essentially statistical. Random variations in time are not considered; these fluctuations are averaged out. Randomness is introduced into the system models by considering basic parameters such as resonance frequencies and observation position to be selected statistically. Simplifying assumptions on the damping and mode shapes are made. The average and variance of power injected by point sources are calculated. The statistics of response near and away from the driving point are also found. It has not been possible to calculate the exact forms for the response distributions. Accordingly, in order to find confidence coefficients for estimation intervals, a distribution is chosen ad hoc. The selected one, the gamma distribution, has several desirable features. It is, in fact, exact for some important cases. Two kinds of estimation intervals are derived and applied to some simple examples. Finally, an interesting alteration of the frequency‐spacing statistics, inspired by nuclear spectroscopy, is explored. It is found that a “level‐repulsion” phenomenon causes small separations in resonance frequency to be less probable. This can smooth the multimodal response of systems in some important, practical instances.

Journal ArticleDOI
TL;DR: In this article, it was shown that at sufficiently low frequencies (i.e., below 100 kHz for water), the observed vaporous cavitation threshold depends primarily upon the initial growth of a vapor cavity in an imperfectly wetted crevice.
Abstract: Galloway, Strasberg, Barger, and others have observed that vaporous cavitation thresholds for unfiltered water at frequencies below 100 kHz depend strongly upon the air content of the water sample; and Strasberg has offered theoretical arguments that are consistent with these observations. Greenspan, on the other hand, has observed little dependence on air content at 30–60 kHz for carefully filtered water and for liquids that readily wet solids. Strasberg's theory has been re‐examined and extended to include the size of solid impurities (i.e., motes) in the liquid sample and the degree to which the liquid wets them. The results appear to explain the above observations, as well as some temperature and frequency effects, and are entirely consistent with Harvey's basic hypothesis that gas cavities trapped in surface crevices of motes are the primary source of cavitation nuclei. It is concluded that, at sufficiently low frequencies (i.e., below 100 kHz for water), the observed vaporous cavitation threshold depends primarily upon the initial growth of a vapor cavity in an imperfectly wetted crevice. [This work is supported by the Office of Naval Research.]

Journal ArticleDOI
TL;DR: Reflection of plane waves from stress free flat surface of micropolar elastic half space, presenting reflection laws and amplitude ratios as mentioned in this paper, and amplitude ratio of plane wave reflection laws.
Abstract: Reflection of plane waves from stress free flat surface of micropolar elastic half space, presenting reflection laws and amplitude ratios

Journal ArticleDOI
TL;DR: In this article, a method for increasing this damping by cutting the constraining layer into appropriate lengths is discussed, based on effective complex elastic moduli of an equivalent homogeneous medium.
Abstract: Viscoelastic materials are used extensively to damp flexural vibrations of metallic structures; it has been known for some time that the energy dissipation due to shear strain in the viscoelastic layer can be increased by constraining it with a stiffer covering layer. In this paper, we will discuss a method for increasing this damping by cutting the constraining layer into appropriate lengths. The analysis for a single layer of this treatment is relatively straightforward. The damping can be increased still further by using several layers; in this case, the analysis is based upon effective complex elastic moduli of an equivalent homogeneous medium. One result found from this analysis is that, if the element length of the constraining layer is optimum, the damping depends primarily upon the stiffness of the constraining layer and the loss coefficient of the viscoelastic material, and only indirectly on the shear modulus of the viscoelastic layer. Experimental data is presented for comparison with the theoretical predictions.

Journal ArticleDOI
TL;DR: This digital simulation method is useful to “pre‐audit” architectural designs before construction and to investigate subjective correlates of a wide variety of reverberation processes.
Abstract: Digital computers, through their capability for accurate simulation of complex phenomena, have permitted new insights into a number of important problems arising in sound transmission in reverberant spaces. (1) Starting with reverberation‐free speech or music signals as inputs, computers can add echoes and reverberation with specified delays, spectral content, and decay characteristics. The computer produces several output signals that—when radiated from loudspeakers in an anechoic chamber—produce, at a listener's ears, sound‐pressure waves resembling those in real halls. To ensure “externalization” and proper directions of echo arrivals, the computer program is based on the measured sound diffraction around the listener's head. This digital simulation method is useful to “pre‐audit” architectural designs before construction and to investigate subjective correlates of a wide variety of reverberation processes. (2) Digital computers have made possible the simulation of frequency and space response of stationary sound fields and the calculation of their statistical properties. These properties are important for the design of electroacoustic systems and the evaluation of measurements in reverberant enclosures. (3) Reverberation theories, based on the geometrical acoustics, are being refined by ray‐tracing studies on digital computers. These studies have revealed significant discrepancies in existing reverberation‐time formulas and unexpectedly large dependencies of the decay rate on the shape of the enclosure and the distribution of sound‐absorbing materials.

Journal ArticleDOI
TL;DR: This study was an attempt to account for the motor control of speech production by a model in which discrete phoneme commands are modified according to phonological context by three motor system mechanisms.
Abstract: This study was an attempt to account for the motor control of speech production by a model in which discrete phoneme commands are modified according to phonological context by three motor system mechanisms. The model was evaluated by consideration of high‐speed cinelluorograms, and electromyograms from nine articulatory locations, recorded while one subject produced 36 consonant‐vowel consonant monosyllables. The syllables were formed by every possible combination of initial and final consonants /b/, /d/, and /g/, and the syllable nuclei /i/, /u/, /ae/, and /ɔ/. In every possible case, some aspect of the motor control of a later syllable component was influenced by the identity of the previous one. Except in a few cases, some aspect of the motor control of an earlier syllable component was influenced by the identity of the following one. These latter influences were of greater magnitude and complexity, and more reflected in movement, in the initial consonant than in the vowel. Some of the context effects on phonemes could be accounted for by the three motor system mechanisms but a number could not. The results suggested that syllabic factors are influential in the “premotor” command structure of speech, and, in particular, that the CV form is a relatively cohesive component of CVC syllables.

Journal ArticleDOI
TL;DR: In this article, the authors report the results of tests of the ability of an observer to estimate the distance of speech signals that originate, or appear to originate, at the intersection of the horizontal and median planes in anechoic space.
Abstract: This paper reports the results of tests of the ability of an observer to estimate the distance of speech signals that originate, or appear to originate, at the intersection of the horizontal and median planes in anechoic space. Both live and recorded sources were used over a range of distances from 3 to 30 ft. For such a range (i.e., where differences in the selective effect of air absorption with frequency are relatively small), the results showed a wide difference in observer ability to estimate the distance for these two types of sources. When loudspeaker sources of recorded speech were employed, the judgments reported were essentially independent of the actual distance involved. This was true whether single, multiple symmetrical, or asymmetrical arrays were employed. When a live voice was used as a source, a rather marked degree of ability to estimate relative distance was found depending on the type of vocal output employed and on the degree to which normal level changes with distance were eliminated as a parameter. Use of a shouted voice resulted in overestimating the distance, whereas the apparent distance was foreshortened when a whispered voice was employed.

Journal ArticleDOI
TL;DR: If each click is assumed to produce identical, linearly superposing CEF's, then the relative amplitudes of adjacent peaks of the CEF can be determined by this nulling technique when k=1, and the results obtained do not agree and are inconsistent with the linear superposing technique.
Abstract: The interleaving peak structure of poststimulus time (PST) histograms of low‐frequency cochlear‐nerve‐fiber responses to acoustic clicks of both polarities implies that a cochlear neuron is excited by a damped oscillatory waveform, referred to here as a click excitation function (CEF) that is half wave rectified. For a stimulus consisting of two closely spaced clicks, the amplitude and delay of the second click relative to the first click can be adjusted so that one‐half cycle of the CEF due to the first click nulls or cancels a particular half‐cycle of the other CEF. A null produced in this manner can be detected from the PST histograms of responses of a particular fiber to both polarities of the click‐pair stimulus. This nulling technique has been used to study the responses of cochlear nerve fibers to combination click stimuli. Details of these measurements are given, and the interpretation of the results leads to the consideration of two apparently different nonlinear phenomena. These results and the techniques used to obtain them provide a means by which certain constraints on the mechanisms of the peripheral auditory system can be evaluated.

Journal ArticleDOI
TL;DR: Applying a digital computer both for signal generation and response processing, some experiments were carried out in which comparisons were made of the timbres of complex tones, finding that the timbre difference between a tone consisting of only sine or cosine terms and atone consisting of alternative sine and Cosine terms represents the maximal possible effect of phase on timbre.
Abstract: Applying a digital computer both for signal generation and response processing, some experiments were carried out in which comparisons were made of the timbres of complex tones. The tones had equal loudness and pitch but different phase and amplitude patterns of the harmonics. The stimuli were presented successively in triads, and the subject's task was to select the most similar and most dissimilar pairs. The results were analyzed with Kruskal's multidimensional‐scaling program. In general terms, the most important findings are: (1) the timbre difference between a tone consisting of only sine or cosine terms and a tone consisting of alternative sine and cosine terms represents the maximal possible effect of phase on timbre; (2) the maximal effect of phase on timbre is quantitatively smaller than the effect of changing the slope of the amplitude pattern by 2 dB/oct and is less for higher than for lower frequencies; (3) the effect of phase on timbre appears to be independent of the effect of amplitude pattern and of the loudness factor.

Journal ArticleDOI
TL;DR: In this paper, exact solutions of the equations of the fully coupled linear theory of piezoelectricity are obtained for some simple types of two-dimensional waves in an infinite plate.
Abstract: Exact solutions of the equations of the fully coupled linear theory of piezoelectricity are obtained for some simple types of two‐dimensional waves in an infinite plate. It is shown that the coupling of the mechanical and the electrical fields can give rise to dispersion curves with complex branches and to waves that are largely confined to the region near the major surfaces of the plate.

Journal ArticleDOI
TL;DR: The data reveal that, although stop cognates are similar in the characteristics of the constrictory articulation, the voiced stop is produced with a larger supraglottal volume than its voiceless cognate.
Abstract: Cinefluorographic films and throat microphone recordings were obtained from three male speakers for 18 pairs of utterances in which one member of a pair differed from the other only in a stop cognate, i.e., /p, t, k/:/b, d, g/. The data reveal that, although stop cognates are similar in the characteristics of the constrictory articulation, the voiced stop is produced with a larger supraglottal volume than its voiceless cognate. The volume differences are caused by a lengthening and expansion of the oropharynx during voiced stops. These processes, which probably satisfy the aerodynamic requirements of voicing, are interpreted to be the results of muscular action rather than passive responses of the vocal tract. Further observations concern the articulatory dynamics of consonant production and may be relevant to the description of neuromotor commands in speech production.

Journal ArticleDOI
TL;DR: The results are compared with some previous work on interaural discrimination and time‐intensity trading and with the prediction of the EC model relatinginteraural time discrimination to interaurally amplitude discrimination.
Abstract: Preliminary experiments were performed on the discrimination of interaural time and the discrimination of interaural amplitude for lateralization images both on and off the midline, using a common set of subjects and a common experimental configuration. The results are compared with some previous work on interaural discrimination and time‐intensity trading and with the prediction of the EC model relating interaural time discrimination to interaural amplitude discrimination.

Journal ArticleDOI
TL;DR: The results show that subjects are much more sensitive to changes in vowel duration than toChanges in consonant duration, and that changes in segment duration may have several different perceptual effects, including changes in perceived stress and perceived rhythm.
Abstract: A threshold‐tracking method was used to measure both the incremental and the decremental just noticeable differences for segment duration in naturally spoken sentences. The measurements were made for a /p/ in five different contexts, including two that were word initial, two that were word medial, and one that was word final; for /∫,m/, and /l/ all in initial prestress position; and for a stressed vowel /ɔ/. The results show that subjects are much more sensitive to changes in vowel duration than to changes in consonant duration, and that changes in segment duration may have several different perceptual effects, including changes in perceived stress and perceived rhythm. When subjects based their judgments on changes in perceived stress or rhythm, they were usually able to detect smaller changes in duration than when they attended to other aspects of the stimuli.