bluuebunny/crossref_metadata_2025 huggingface dataset:
Speech Analysis and Synthesis by Linear Prediction of the Speech Wave
Authors: Atal B. S.
Published: October 2005

A method of representing the speech signal by time-varying parameters relating to the shape of the vocal tract and the glottal-excitation function is described. The speech signal is first analyzed and then synthesized by representing it as the output of a discrete linear time-varying filter, which is excited by a suitable combination of a quasiperiodic pulse train and white noise. The output of the linear filter at any sampling instant is a linear combination of the past output samples and the input. The optimum linear combination is obtained by minimizing the mean-squared error between the actual values of the speech samples and their predicted values based on a fixed number of preceding samples. A 10th-order linear predictor was found to represent the speech signal band-limited to 5kHz with sufficient accuracy. The 10 coefficients of the predictor are shown to determine both the frequencies and bandwidths of the formants. Two parameters relating to the glottal-excitation function and the pitch period are determined from the prediction error signal. Speech samples synthesized by this method will be demonstrated.
Inadequacy of a Spectral Description in Relationship to Speech Perception
Authors: Blesser B. A.
Published: October 2005

A new experimental technique, in which the spectrum of natural speech undergoes a rotation so that the high-frequency energy becomes low frequency and vice versa, has been used to demonstrate that spectral distortion, unlike amplitude distortion and filtering, creates a speechlike signal, which, although not initially intelligible, can be learned after about 4 h of practice. Pairs of subjects placed in the medium learned to understand each other by attempting to converse. The content of their conversations was analyzed for learning strategies, and a model of learning stages was generated. An extensive series of tests was given at the end of each practice session in order to gain insight into the perceptual relevance of phonemes, words, syntax, and semantics. The test results showed that a subject's performance on identification and discrimination of isolated speech elements did not predict comprehension of sentences. Moreover, there were strong indications that the prosodic features, which were unaffected by the transformation, and the use of metalinguistic cues (syntax, semantics, etc.) compensated for the loss of reliable spectrally dependent phonetic cues. [Supported by NIGMS grants.]
Audibility Curve of the Common Canary
Authors: Dooling R. J. , Mulligan J. A. , Miller J. D.
Published: October 2005

The auditory sensitivity of one type of common canary (Belgian Waterschlager) was measured by behavioral audiometry. Four birds were trained by instrumental avoidance conditioning in a double-grill cage and then their thresholds for pure tones (0.25–9.0 kHz) were measured. Auditory sensitivity was maximum from 2.0 to 4.0 kHz and sharply declines for frequencies above or below this region. Indeed, the sensitivity curve appears to be matched to the average spectrum of this bird's songs and calls.
Sensitivity of the Green Sea Turtle's Ear as Shown by Its Electrical Potentials
Authors: Ridgway Sam H. , Wever E. G. , McCormick James G. , Palin Jery , Anderson John
Published: October 2005

Both aerial and bone-conduction cochlear-potential runs demonstrated similar frequency sensitivity curves. Maximum sensitivity was in the 400-Hz region, dropping off rapidly above and below. A point-source bone-stimulation cochlear-potential study of the Green Sea Turtle's head showed that the most sensitive receiving area is over the head of the columella.
Function of the Porpoise Ear as Shown by Its Electrical Potentials
Authors: McCormick James G. , Wever E. G. , Ridgway Sam H. , Palin Jerry
Published: October 2005

The function of the porpoise ear was studied with cochlear potential experiments on 15 anesthetized Tursiops truncatus and Lagenorhynchus obliquidens. By performing surgical manipulations during cochlear-potential recordings, the following was found: (1) The external meatus of the porpoise is no more important for sound conduction to the ear than is the surrounding tissue of the head. (2) The ear drum and tympanic ligament are not necessary for the transmission of sound energy to the inner ear. A point-source bone stimulation cochlear-potential map of the body revealed that (1) the skull of the porpoise is acoustically isolated from the ear; and (2) the lower jaw bone is an effective transducer of sound energy to the ear when made to vibrate with displacement towards the approximate axis of the stapes. Underwater cochlear-potential audiograms demonstrated good sensitivity for the porpoise at least as high as 100 kHz.
Phase Dependence of Two-Tone Inhibition in Frog Auditory Nerve Fibers
Authors: Liff Harold
Published: October 2005

Frog auditory nerve fibers that innervate the amphibian papilla have complex coding characteristics. Single unit recordings show excitation by low-frequency tones that can be suppressed by simultaneously presenting a properly chosen “inhibitory” tone of higher frequency. We have studied this inhibitory phenomenon using two-tone stimuli in which the inhibitory tone was a harmonic of the excitatory tone, and the phase difference between the two was controlled. Our results show that the responses to such stimuli are strongly dependent upon the phase difference. At low inhibitor intensities (relative to the excitor intensity), inhibition was observed for a range of phase differences, and no inhibition was observed outside of this range. As the intensity of the inhibitor was increased, the range of phase differences for which inhibition was seen widened, and, at sufficiently high inhibitor intensities, all phase differences caused inhibition. [Research supported in part by the U. S. Air Force Office of Scientific Research and the National Institute of Neurological Diseases and Stroke.]
On-Line Processing—A Modular Approach
Authors: Kaufman Myron , Kasper Frand
Published: October 2005

Several self-contained instruments used to obtain on-line analysis of random and deterministic signals, are discussed. A real-time spectrum analyzer using Metal Oxide Semiconductor Field Effect Transistor Ensemble (MOS FET) shift-registers storage provides real-time analysis of transient and continuous data. This controllable circulation-rate storage permits optional features such as transient-time-function plotting, a choice of true constant-percentage or constant-bandwidth analysis. Analysis of 20 000 Hz are handled with a dynamic range up to 54 dB. Analysis at higher frequencies is accomplished by frequency translation extending the range to 100 000 Hz with resolutions as fine as 0.5 Hz. Ensemble averaging provided by a separate device uses an identical storage medium for efficient provisioning. As many as 1024 independent spectra may be averaged. A digital correlation and probability analyzer is described providing a 100-point autocorrelation, cross-correlation, or probability function with a selectable sampling rate and predetermined integration time. Compatibility of instruments makes possible, for example, the autocorrelation function of a one-shot transient captured in the analyzer storage or the Fourier transform of the cross-correlation function.
Real-Time Spectrum Analysis in the Field Using a General Purpose Computer
Authors: Fitzgerald E. M. , Petrini A. A. , Zimmerman J. K.
Published: October 2005

Recent development of new hardware and software techniques is discussed. The application of the small general-purpose computer with specialized software in conjunction with limited analog hardware is used to provide experiments with real-time results. Additional benefits are: automation of calibration and data taking, data output in the form of plots and CRT display, and flexibility to adapt to varying tasks. A statistical sampling technique is employed to provide accurate 13-oct-band levels up to 100 kHz. The same equipment configuration is used to perform band-level analysis, narrow-band FFT analysis, cross correlation, averaging and standard deviation of processed results, and comparison of test results to a reference or to each other. Uses in the field to perform underwater acoustical measurements and aircraft-noise measurements are illustrated. Corrections to standard-day conditions and real-time computations of effective perceived noise level for jet aircraft fly overs are presented. A conversational-mode programming language for signal processing of acoustical data is described.
Air-Borne and Solid-Borne Noise Discrimination of a First-Order Gradient Microphone Based on the Foil-Electret Principle
Authors: West J. E. , Sessler G. M.
Published: October 2005

A first-order pressure-gradient microphone, which discriminates against air-borne noise as well as against solid-borne vibrations transmitted through the microphone case, is described. The microphone is based on the electrostatic principle and utilizes a foil-electret diaphragm [G. M. Sessler and J. E. West, J. Acoust. Soc. Amer. 40, 1433 (1966)]. For air-borne sound originating from a spherical sound source positioned close to the microphone, the frequency response is flat within ±3 dB in the frequency range 0.1–3 kHz with a sensitivity of about −50 dB re 1 V/μbar in the direction of the microphone axis. In a plane sound field, the sensitivity is lower (by 20 dB at 0.3 kHz) and rises proportional to frequency over most of the mentioned range. The microphone therefore discriminates against plane-wave noise fields in close-talking applications. Rejection of solid-borne vibrations is due to the lightness of the foil-electret membrane. Compared with electrodynamic and electromagnetic transducers, the foil-electret microphone responds 30–40 dB less to solid-borne vibrations referred to equal response for air-borne sound.
Undirectional, Underwater-Sound, Pressure-Gradient Transducer
Authors: Marciniak R. D.
Published: October 2005

Results are presented for the design, construction, and calibration of an experimental, low-frequency, pressure-gradient, underwater sound transducer that utilizes a liquid-filled, acoustic phase-shift network to obtain a unidirectional (cardioid) directivity response over a prescribed frequency range. A solution for satisfying a number of analytical relationships, which are known to exist among certain physical parameters of the device, its environment, and the acoustic phase-shift network, is outlined. Design specifications defining the over-all physical configuration of the transducer, the acoustic phase-shift network parameters, and the piezoceramic pressure-gradient element, are given. Presentation of experimental data, in the form of azimuth directivity responses, demonstrates that a unidirectional (cardioid) response was achieved consistently between 4000 and 6000 Hz. Front-to-back discriminations ranging between 15 and 27 dB were recorded over the upper half of the frequency range (5000–6000 Hz), while 10- to 15-dB discriminations were recorded over the lower half (4000–5000 Hz) of the frequency range. In addition, a nominally flat free-field voltage response (M0) of −85 dB +2 dB re 1 V//μbar was achieved over the frequency range of 4000 to 6000 Hz.
Mathematical Model for the Class II Flextensional Underwater Acoustic Transducer
Authors: Boone J. N. , Royster L. H. , Nelson R. A.
Published: October 2005

A mathematical model for axisymmetric free vibrations of a segmented model of the Class II Flextensional Underwater Acoustic Transducer Shell is developed. Digital computer solutions for the matrix eigenvalue problem are discussed. For the case of short cylindrical end sections, the eigenvalues and eigenvectors obtained from the approximate model approached those obtained utilizing a previously developed curved-beam model for horizontally guided boundary conditions. By extending the analysis to include the piezoelectric stack, a fundamental resonance frequency was obtained that is within 5% of published experimental results. [Work was sponsored by the Office of Naval Research, Acoustics Programs, and by the Naval Underwater Sound Laboratory.]
On the Significance of Coupled Solutions in Vibration Analysis of Piezoelectric Cylindrical Shells
Authors: Kalnins Arturs , Drumheller Douglas S.
Published: October 2005

This paper is concerned with the vibration analysis of ferroelectric ceramic hollow cylindrical shells that are often used as elements of electromechanical transducers. The analysis of such elements can be performed in an approximate (or uncoupled) way, so that, by violating some of the constitutive relations, either a constant electric field or a constant electric displacement is assumed between electrode surfaces. In the former case, the reasonance frequencies are the same as those of an elastic material, with the elastic parameters measured at a constant electric field, while in the latter case, the antiresonance frequencies are the same as those of an elastic material, with the elastic parameters measured at a constant electric displacement. A coupled piezoelectric-elastic analysis, where all the electrostatic and constitutive relations are satisfied with the same accuracy as in a shell theory, shows that the resonance and antiresonance frequencies lie between those predicted by the two uncoupled cases. Such a shift in the frequencies is important, because the electromechanical coupling coefficient is proportional to the distance between the resonance and the antiresonance frequencies. It is concluded that the coupled analysis will predict lower coupling coefficients than the uncoupled analysis.
Optical Probing of Acoustic Surface Waves
Authors: Lean E. G. H. , Powell C. G. , Pole R. V.
Published: October 2005

Light diffraction by acoustic surface waves is used as a probe to study the propagation characteristics of Rayleigh waves on y-cut z-oriented LiNbO3. Both the surface deformation and the periodic variation of the index of refraction in the medium, which diffracts the transmitted light beam, are related to the particle displacements of the propagating Rayleigh waves. For the reflected beam, only the surface deformation contributes to the diffracted light. Experimental demonstrations of using the optical probe to observe the Fresnel diffraction of a rectangular interdigital transducer to measure the transmission and reflection coefficients of a transducer and to determine the propagation characteristics, such as velocity, attenuation of Rayleigh waves on a LiNbO3 substrate and on a metalized LiNbO3 substrate, will be discussed.
Optical Observation of Conical Reflection of Ultrasound from Liquid-Solid Boundaries
Authors: Diachok Orest I. , Mayer Walter G.
Published: October 2005

Light diffraction and schlieren techniques are used to investigate the behavior of ultrasonic waves incident on a liquid-solid flat interface. It is shown that if the incidence is at the Rayleigh angle, the reflection from an isotropic solid takes place into a cone rather than only in the forward direction. The reradiation patterns from a single-crystal copper sample is found to be governed by crystal symmetry and the Rayleigh and pseudosurface-wave velocities. [This work was supported by the Office of Naval Research, U. S. Navy.]
Mapping of Ultrasonic Fields Using Liquid Crystals
Authors: Cook Bill D. , Werchan Ronald
Published: October 2005

The temperature-sensitive properties of certain cholesteric liquid crystals permits their use for mapping ultrasonic fields. The liquid crystalline material is sprayed on a thin plastic or rubber sheet, which absorbs a fraction of the ultrasound. The liquid crystals reversibly exhibit iridescent color effects at various temperatures while in transition from solid to liquid state. The thermal properties of the substrate controls to large degree the sensitivity and resolution of this system. Mappings of ultrasonic fields from typical transducer configurations are presented along with data on sensitivity. [Supported by the Office of Naval Research.]
Development and Evaluation of a Noise with Speech-Envelope Characteristics
Authors: Horii Y. , House A. S. , Hughes G. W.
Published: October 2005

A noise whose amplitude envelope followed closely that of a speech signal was generated by multiplying white noise and the amplitude envelope of the speech. When the original speech and the derived noise are added together after appropriate alignment, the signal-to-noise ratio (S/N) of the combined signals is virtually constant, that is, is virtually non-varying on a short-time basis. Articulation functions were determined in such noise and in continuous white noise (where S/N varies on a short-time basis). Within the range of S/N studied, the gains of the functions were the same in both kinds of noises, being 2.5%/dB for consonants and 4%/dB for vowels. Differential effects on speech intelligibility and response stability are discussed, together with advantages and disadvantages of this method of masking speech. [This work was supported in part by the Air Force Cambridge Research Laboratories, Office of Aerospace Research.]
Laterality Factors in the Perception of Sentences Varying in Semantic Constraint
Authors: Pisoni David B. , Jarvella Robert J. , Tikofsky Ronald S.
Published: October 2005

The present experiment explored the differential rôle of the left and right ears in perceiving linguistic stimuli on the level of sentences. Sentences with identical syntactic structure but varying in semantic constraint [i.e., “The farmer plowed the field” (semantically well integrated) vs “The Indian hit the table” (semantically poorly integrated)] were presented to the left or right ear of 10 right-handed subjects. Listeners were required to listen to and then immediately repeat each sentence after presentation. A special speech-noise distortion circuit developed at the Communication Sciences Laboratory of The University of Michigan [M. A. H. O'Malley and G. E. Peterson, “An Experimental Method for Prosodic Analysis,” Phonetica 15, 1–13 (1966)] was used to produce a masking signal that provided a constant signal-to-noise ratio for the entire utterance. The circuit derives the speech masking signal from the utterance itself by destroying the segmental information although maintaining the instantaneous power spectrum and pitch information of the sentence. Masking output of the circuit was delivered to both ears of a listener while the speech signal was presented to only one ear at a time. This procedure provided an opportunity to examine the effect of speech distortion on the rôle of the two ears in perceiving sentences. Preliminary results indicate significant differences in immediate recall for ears—the right ear being more efficient than the left ear. Results of this pilot study indicate that: (1) differential ear preference is a function of cerebral dominance; and (2) there are differences between the two hemispheres in processing different types of linguistic information on the sentence level.
Target Simulation and the Sonar of Bats
Authors: Simmons James A.
Published: October 2005

The acoustic emissions of an echolocating bat can be reproduced and fed back to the bat using condenser microphones, small electrostatic loudspeakers, and appropriate electronics. Such a system simulates the presence of target objects for the bat's sonar. The individual acoustic parameters of simulated echoes are under better experimental control and are easier to manipulate than are the parameters of echoes from real targets. The acuity of resolution of short time intervals by echolocating bats has been measured with simulated echoes in simultaneous discrimination experiments. Two different species, Eptesicus and Phyllostomus, can process and identify echoes that differ in arrival time by as little as about 60 μsec. For both species, these time measurements are in good agreement with previous measurements of the limits of range resolution, approximately 12 mm. In addition to its application in the study of auditory cues for range perception by bats, the simulator is being used to measure the bat's acuity in discriminating fine spectral differences between echoes. Eventually, we may identify the auditory cues that provide the bat with signatures for target shape and orientation.
Sound Pressure at the Entrance of the External Meatus as a Function of the Azimuth of a Tone
Authors: Harrison J. M. , Downey P.
Published: October 2005

Sound pressure of tones of 4, 10, and 20 kHz was measured at the external meatus as a function of the azimuth of the source. The measurements were made under free-field conditions in man, squirrel monkey, bat, and rat. At 4 kHz, rat and man showed consistent variations of sound pressure as a function of azimuth. Interaural intensity differences were estimated to be sufficient to support localization of a tone at a minimum angle of 2.1° (rat) and 1.13° (man) from the midline. No regular sound pressure differences were obtained in bat and squirrel monkey. At 10 and 20 kHz, rat, bat, and monkey showed consistent variations of sound-pressure level with azimuth of at least 15 dB. Interaural differences were sufficient to support localization of a 10-kHz tone at 1.13° (monkey), 2.0° (bat), and 1.5° (rat). Similar results were obtained at 20 kHz. No measurements were made on man at 10 and 20 kHz. [Supported by the National Science Foundation.]
Acoustical Vestibular Stimulation in the Guinea Pig: Vestibular Nerve Response
Authors: Parker D. E.
Published: October 2005

Responses from single vestibular nerve cells were elicited by slow time-varying pressure changes in the external auditory meatus. The resting discharge from these cells was regular and rapid (80–160 pulses/sec). Stimulation generally produced a reduction of discharge rate for those cells that could be acoustically activated (approximately 40%). Minimum stimulus intensities required to elicit clear responses were 1–2 cm Hg and response latencies ranged from 1 to 3 sec. The decay in discharge rate appears exponential with a time constant in the area of 2 sec. Data from the present study correlate well with previously reported observations on head and eye movements elicited by acoustical stimulation. Minimum stimulus intensities required to demonstrate head and eye movements were approximately 2 cm Hg and response latencies ranged from 2 to 8 sec. The observations obtained to date strongly support the hypothesis that acoustical stimulation may activate vestibular receptors. Moreover, these data suggest that the mechanism of stimulation may involve a slow flow of fluid through the membranous labyrinth. [Research supported by Aerospace Medical Research Laboratories, USAF.]
Effect of Coatings on the Acoustical Properties of Materials Measured in an Impedance Tube
Authors: Eriksson L. J. , Higgs R. W. , D'amico P. M.
Published: October 2005

The change in the measured sound velocity caused by coatings on materials placed in impedance tubes is examined from an experimental and theoretical viewpoint. Assuming a rigid tube surrounding a cylindrical sample of infinite length coated with a material with a small, but finite, shear modulus, a dispersion relation is obtained. The effect of changes of the sample radius and the wavelength on the velocity is studied with the aid of a digital computer. The asymptotic values of the modes are identified. These theoretical results show that samples coated with materials whose sound velocity is less than that of the sample have a reduced sound velocity. The sound velocity was measured for a typical plastic surrounded by a water capillary, and the results were compared to the above theory for various sample diameters.
Acoustic Maintenance Aids
Authors: DiRita R. A.
Published: October 2005

By means of monitoring audible/vibrational signals, two methods have been developed that make it possible to detect, on an incipient basis, the failure of a component, mechanism or machine. The first approach involves real-time monitoring of a system via permanently installed transducers. As part of this scheme, a computer system is used to analyze the transducer signals so as to provide visual or audible warnings that a malfunction might possibly occur. The second technique revolves around a specially developed frequency/amplitude detection unit capable of scanning a transducer signal over a very wide frequency/amplitude band. The feature that makes it unique is its ability to classify the amplitude at any one frequency as being indicative of normal, marginal, or defective operation. Both of the methods were developed to compensate for the inadequacies that present-day maintenance technology provides. While the concept of signature recognition as a maintenance aid is not new, the specific techniques developed are unique and suggest an approach to failure prevention that might be particularly attractive to users of data processing equipment.
Analysis of the Liquid Surface Motion in Liquid Surface Acoustical Holography
Authors: Bander T. J. , Hildebrand B. P.
Published: October 2005

A linear partial differential equation is obtained that describes the motion of a liquid surface when impinged upon by pulsed ultrasonic waves. The case of two pulsed plane ultrasonic waves incident upon the liquid surface over the same area at equal but opposite angles is treated. A particular solution of the equation is obtained and separated into a ripple displacement superimposed on a uniform displacement, both of which oscillate. Numerical values of frequencies of oscillation and displacements are obtained for various liquids. The theoretical analysis is compared with experimental results. In addition, the transfer function relating the liquid surface displacement to the acoustic radiation pressure, and the scattering coefficient, for the light reflected from the liquid surface as a function of surface displacement, are obtained.
Optical Discrimination between Two Mechanisms in Acoustically Birefringent Liquids
Authors: Riley Ward A. , Klein W. Richard
Published: October 2005

Two mechanisms of acoustically induced optical anisotropy in liquids have been advanced. For small molecular size, Peterlin has attributed the effect to molecular alignment by the dynamic velocity gradient of the acoustic field. For large molecules, Oka introduced the “Rayleigh Disc” mechanism. The former leads to an induced optical anisotropy in the liquid having the same spatial and temporal periodicity as the acoustic field. The latter, since it depends quadratically on the acoustic pressure gradient, has half the spatial periodicity of the acoustic field. Thus, the two mechanisms lead to distinctly different optical effects. By studying the polarization of the optical far-field diffraction pattern produced by the ultrasonic field, it is possible to discriminate between mechanisms and study the effect due to either mechanism.
Parametric Excitation of Ultrasonic Waves
Authors: Adler Laszlo , Breazeale M. A.
Published: October 2005

It has been reported [J. Acoust. Soc. Amer. 46, 94(A) (1969)] that the observed fractional harmonics in a liquid-filled resonant ultrasonic cavity are parametrically excited ultrasonic waves. This excitation is produced by instabilities introduced through the vibration of the driver transducer that forms one end of the cavity. The theoretically derived threshold criterion that defines the range of amplitudes for such excitation is now tested experimentally. Since the theory predicts that the transducer voltage at threshold is a function of the driver frequency, the absorption per wavelength of the medium, and the cavity length, a change in either of these parameters causes a corresponding change in the transducer voltage at threshold. There is reasonable agreement between theory and experiment. To develop the model further, a nonlinear equation of parametric action is derived. The solution of the equation including the first nonlinear terms confirms the experimental observation that the nonlinearity of the medium does not affect the magnitude of the threshold for parametric oscillation. [Research sponsored by the U. S. Office of Naval Research.]
Speaker-Machine Interaction in a Limited Speech Recognition System
Authors: Makhoul John I.
Published: October 2005

The performance of a speaker-independent limited speech recognition system for adult male and female speakers is examined. In cases of incorrect recognition, on the basis of results, the speaker is instructed to change his articulation to effect correct recognition. Methods of changes in articulation and the effects they have on recognition performance are discussed. The vocabulary used consists of 55 /əCVd/ syllables, where C is one of six stop and five fricative consonants, and V is one of five tense vowels. The recognition depends on the extraction of several acoustic features. The positions of spectral energy concentrations are determined and their relative movements are tracked, especially along the CV transitions. [This work was supported in part by grants from the National Institutes of Health.]
Pitch Determination by Measurement of Harmonics. II. The Hipex System
Authors: Miller Ralph L.
Published: October 2005

Work on the method of pitch determination by measurement of harmonics described previously [J. Acoust. Soc. Amer. 44, 390 (1968)] has been continued to determine its performance characteristics in relation to system parameter changes. The resulting experimental arrangement has been termed the Hipex (Harmonic Identification Pitch Extraction) system. A statistical study of “period histograms” with a wide range of pitches gives an optimum adjustment for the peak detection process needed to identify the fundamental period. Percentage errors are used for comparative purposes in assessing the performance for different numbers of harmonic measurement channels as well as for different types of speech. Characteristic error distributions lead to simple logic for error elimination. Performance of the Hipex system in the presence of noise is unusually good. Usable pitch information has been obtained from speech in an approximately zero signal-to-noise level. The effectively coherent addition of harmonic signals in the period histogram regardless of their phase in the speech wave implies that the technique is approaching an optimum for determination of the fundamental period from the available information.
Adaptive Delta Modulation of Speech with a One-Bit Memory
Authors: Jayant N. S.
Published: October 2005

We define an adaptive delta modulator in which the adaptation of the step size mr at the rth sampling instant depends on the comparison of two channel symbols—the bits Cr and Cr−1 corresponding to the rth and (r−1)st sampling instants. Specifically, the ratio of mr to the previous step size mr−1 is +P or −Q depending on whether Cr and Cr−1 are equal or not. We indicate the step response of the delta modulator and present results on the simulation of the adaptation logic on a speech sample that was band limited to 3.3 kHz. On the basis of signal-to-error ratios, we note that the equations PQ = 1 and P = 1.5 define optimal adaptation characteristics for sampling frequencies of 20, 40, and 60 kHz, and also that PQ⩽1+ε(ε≪1) represents a strong condition for stability. Finally, we comment on the relative performance of the adaptive delta modulator and of logarithmic PCM, and find that at a 60-kHz sampling rate, the delta modulator achieves a 7 log-bit PCM quality.
Temporary-Threshold-Shift Recovery from Impulse and Steady-State Noise Exposure
Authors: Fletcher John L.
Published: October 2005

Fifteen young male subjects with hearing levels in the range of 125–8000 Hz no worse than 25 dB ISO were exposed to impulse and continuous noise separately until a criterion temporary threshold shift (TTS) of 25 dB at any of the frequencies monitored was reached. Recovery from the TTS was then observed at frequencies from 3–14 KHz for time intervals of 15 min, 30 min, 1, 2, 4, and 7 h postexposure. Recovery curves from impulse-noise-induced TTS differed significantly from the continuous-noise TTS curves, with recovery much slower from impulse TTS. Results indicate the need for further research in this area.
Temporary Changes of the Auditory System Due to Prolonged Exposure to Noise
Authors: Mills J. H. , Gengel R. W. , Watson C. S. , Miller J. D.
Published: October 2005

One subject was exposed to an octave-band noise centered at 500 Hz for 48 h at 81.5 dB SPL and on a second occasion for 29.5 h at 92.5 dB SPL. Auditory sensitivity was measured before, during, and following the exposure as was temporal auditory summation, loudness, frequency discrimination, and evoked vertex potentials. After 8–12 h of exposure, the growth of temporary threshold shift (TTS) was asymptotic. Recovery from asymptotic TTS required 2–4 days even though the magnitudes of TTS were only about 10 dB for the 81.5 dB exposure and 27.5 dB for the 92.5 dB exposure. Other test results showed that during and following asymptotic TTS, the time constant of temporal auditory summation was decreased, and recruitment of loudness was delayed and incomplete; however, frequency discrimination and the slope of the input-output function of the N1−P2 potential were unaffected.
A Method for Computer-Controlled Group Audiometry
Authors: Silbiger Herman R.
Published: October 2005

The need often exists to screen for hearing acuity a group of subjects that are to be used in subsequent experiments on auditory discrimination or speech transmission quality evaluation. A method has been developed by which the existing multiple listening facility of a laboratory principally devoted to speech quality measurement was adapted to enable screening audiograms to be given to as many as 22 subjects simultaneously. Subjects listen to the stimuli, which are groups of up to three tone pulses, over modified telephone handsets. Subjects are instructed to respond positively only if they hear a specified number of pulses. This has the effect of forcing a conservative criterion upon the subjects, which reduces response variability. Tests are made at seven frequencies between 250 Hz and 4 kHz. Tone levels are varied from 30 dB hearing level to −5 dB hearing level (re ISO-1964) in 5 dB steps. The subject responds on a keyboard that is connected to an SDS-930 computer. This computer entirely controls the testing sequence, presents the stimuli, records the data, and prints out the hearing levels for each subject at the end of the 10-min testing sequence. Hearing levels are generally within 5 dB of those obtained by conventional individual audiograms.
Simple Rules for Controlling Step Size in Up-Down Testing
Authors: Levitt H.
Published: October 2005

A common problem in up-down testing is the choice of step size. Too large a step size results in imprecise estimation, and too small a step size is inefficient in that many data may be badly placed at the start of a test. A highly efficient rule [Wetherill, J. Roy. Statist. Soc. B.25, 1–48 (1963)] is to use a step size on trial n proportional to 1/(nb), where b is the slope of the response curve at the point to be estimated. Simple approximations to this rule are considered for practical applications that limit the size and number of changes in step size. Difficulties arise if both slope and location are unknown prior to the experiment. Fortunately, the efficiency of the procedure is not critically dependent on the slope constant and for practical purposes a rough estimate of slope will do. The technique has interesting possibilities in forced-choice experiments, where, under certain conditions, estimates of slope may be derived from the location estimates.
On the Frequency Analysis of Mechanical Shocks and Single Impulses
Authors: Broch Jens Trampe , Olesen Hans P.
Published: October 2005

The damaging potential of mechanical shocks and impulses are, in general, more conveniently evaluated from frequency domain descriptions than from time domain descriptions. A frequency domain description of a shock is obtained by Fourier transformation of its amplitude versus time trace and may be made either mathematically from a graphic or photographic record of the shock pulse, or electronically by means of analog frequency analysis. This paper concentrates on the analog type of analysis utilizing ordinary, commercially available, analog equipment. By far, the most frequently used analog type of analysis is based on sequential filtering of a signal recorded on magnetic tape. Such analyses can be made in basically two different ways. One method consists in playing the pulse back once per analyzer filter and squaring and integrating the output from the filter over the period of time that it responds to the pulse. A second method consists in repeating the pulse periodically by means of, for instance, a closed tape loop. The practical implications involved in the utilization of either method are outlined and some measurement results presented. The paper concludes with a brief discussion of the advantages and limitations of the techniques used.
Real-Time Shock-Spectrum Analyzer Employing Time Compression and Synthesizing Undamped Response Characteristics
Authors: Bickel Henry J.
Published: October 2005

A hybrid instrument has been developed that computes the positive, negative, maximax, and residual shock spectrum within 50 msec after the shock waveform is captured in its digital memory. Among the features of the system are automatic transient capture, and on-line operation without the need for tape loops. The operating principle and salient performance characteristics of the novel response-spectrum analyzer are described. The undamped filter characteristic is synthesized by time-domain weighting the input and output of a finite-Q filter. This approach overcomes the stability problems that normally arise when essentially undamped filters are synthesized by analog techniques, and furthermore, it minimizes energy storage in circuit elements.
Interpretation and Meaning of Laboratory Determinations of the Effect of Duration on the Judged Acceptability of Noise
Authors: Parry H. J. , Stephens J. K.
Published: October 2005

Laboratory determinations of the effect of duration on the judged acceptability of noises have produced conflicting, if not confusing, results. Further, there is the question of application of so-called duration corrections to the prediction of community response to vehicle noises. The laboratory experiments are reviewed in detail to provide a firm understanding of the present status of that data. This investigation develops the very strong implication that judging duration in terms of equivalent noise levels is, in reality, a type of cross-modality experiment wherein the subject is simply measuring duration. A review of similar cross-modality studies by S. S. Stevens shows that the subjective sound intensity growth rate for brightness and length is equal to that for duration.
Judged Noisiness of Sounds Containing Multiple Pure Tones
Authors: Pearsons K. S. , Wells R. J.
Published: October 2005

Psychoacoustic judgment tests were conducted in an anechoic chamber to investigate the judged noisiness of multiple pure tones (MPT) spanning 3 oct. Various combinations of MPT were compared in noisiness to shaped broad-band noise alone. Frequency spacing of the MPT complexes were 50, 100, and 150 Hz and were presented both with and without various noise spectra. Other stimuli included MPT with a prominent tone at 1600, 3200, 5000, or 7500 Hz. The judgment-test results indicate that the MPT without a prominent tone were judged 2.5 PNdB noisier than similarly shaped broad-band noise. Several computational procedures were employed in the analysis of the results. Comparisons of these calculations with the subjective judgments are presented.
Supersonic Turbulence-Induced Pressure Fluctuations
Authors: Saunders Herbert , Johnson Richard I.
Published: October 2005

Considerable attention has been focused in recent years upon aerodynamic noise propagated by the turbulence in the boundary layer and weight of high-speed vehicles. The recent advent of small (18-in.–14-in. diam) and fast-responding commercially available acoustic sensors has uncovered new approaches in boundary-layer investigations. This paper includes data from wind-tunnel tests on the oscillating pressures and spectra in the boundary layer at M∞ = 4.0. The most striking feature of these data indicates that acoustic levels during transition are higher than in fully turbulent flow, and this has direct implications to vehicle structure and component design. Another surprising characteristic is the difference in the spectra of tripped and naturally induced turbulent flow under the same free-stream conditions and at the same location on the model. Comparisons are made between acoustic base pressure measurements in subsonic and supersonic flow, and further compared to current prediction method. The paper concludes with a brief discussion of the spectral measurements of pressure fluctuations at high frequencies up to 200 000 Hz with preliminary inference to turbulent-boundary-layer characteristics.
Fluctuating Pressures Induced by Three-Dimensional Protuberances
Authors: Robertson Jack E.
Published: October 2005

A wind-tunnel investigation has been conducted to study the flow field induced by three-dimensional protuberances of transonic Mach numbers. Test results for 2-, 4-, and 8-in. diam cylindrical protuberances is discussed. The range of test variables consisted of protuberance heights from zero to 2 diameters, free-stream Mach numbers from 0.60 to 1.60, and unit Reynolds numbers from 1.5 to 4.5 million per foot. Fluctuating-pressure measurements are presented in the form of over-all rms fluctuating pressure coefficients, pressure spectra, cross-pressure spectra, and narrow-band convection velocities. From static- and fluctuating-pressure measurements and oil-flow patterns, the structure of the perturbed flow field has been defined. One of the most significant findings is that the upstream separated flow field induced by three-dimensional protuberances consists of a complex, multiple-vortex system, which generates fluctuating pressures on order of magnitude greater than those observed in two-dimensional separated flows. The extreme fluctuating pressures encountered within the three-dimensional separated flow field are attributed to the shear interaction of two major vortices within the separated region.
Self-Demodulation of a Pulsed Sinusoidal Acoustic Wave in Water
Authors: Mellen Robert H. , Browning David G.
Published: October 2005

Self-demodulation of a pulsed sinusoidal wave results from the nonlinear propagation of acoustic waves of large amplitude. In plane-wave theory, the time derivative of the envelope function squared increases linearily with distance until primary-wave distortion becomes important. For a collimated source, however, there are three distinct secondary-wave regions: the nearfield or “plane-wave” region; an intermediate region, the remaining collimation distance of the primary wave, where secondary-wave divergence balances its growth; the farfield, where the secondary-wave function becomes the second time derivative and both primary and secondary waves diverge. In the last region, the collimated volume tends to act as an end-fire radiator for the secondary wave.
Degraded Synthetic Speech for Classroom Demonstration
Authors: Reynolds J. , Strong W.
Published: October 2005

An effort to create classroom demonstrations for a course, “Descriptive Acoustics of Music and Speech,” is currently in progress. This paper discusses and presents the results of one part of that effort. The utterance, “Robby will like you, Daddy-oh,” was synthesized after degrading the control parameters for a terminal analog synthesizer in various ways [W. Strong, J. Acoust. Soc. Amer. 41, 1434 (1967)]. One category of degradation involved the elimination of one or more formants from the utterance. Another category involved modification of the fundamental frequency (FO) in various manners: increase and decrease the average FO, increase and decrease amount of FO deviation from its average, constant FO, and so forth. A third category attempted to derive “femalelike” speech from male speech by shifting the formants and FO upward. A 4-mid tape will be presented. [Work supported in part by an HEA V-C Fellowship.]
Speech Synthesis with a Vocal Tract Synthesizer
Authors: Purves B. , Blackett K. , Strong W.
Published: October 2005

Two different methods have been used to interpolate between steady-state configurations for an 18 section synthesizer for consonant-vowel-consonant utterances. One method requires continuity of area and rate of change of area at the steady values. The second method uses the steady values as inputs to a critically damped filter. Several different time constants have been used with both methods. Difficulties exist in synchronizing the subglottal pressure and vocal-cord tension with the tract area. Taped examples will be presented.
Functional Characteristics and Advantages of an On Line, Computer-Controlled Speech Synthesizer
Authors: Dixon N. R. , Maxey H. D.
Published: October 2005

The problems to be overcome in achieving synthesis of a practically unlimited variety of speech under computer control are numerous and difficult. This is especially true if a goal is automatic conversion from essentially standard orthographic specification to the acoustic output. In addressing these problems it is probably advantageous to start with a synthesis system that will allow inexpensive successive approximations to the necessary phonetic events. For this purpose, a manually operated hardware synthesizer configuration is well suited. However, after the grosser phonetic problems are more or less overcome, the limitations imposed by a purely hardware synthesizer system become prohibitive of further development; an on-line computer-controlled synthesis system offers maximum flexibility and added accuracy of parameter control The synthesizing procedures and features of this on-line system are presented and discussed.
An On-Line Computer-Controlled Diphone Synthesis System
Authors: Maxey H. D. , Dixon N. R.
Published: October 2005

Speech synthesis by diphone segmentation consists of the stylization and simulation of the perceptually meaninfgul, dynamic portions of the acoustic continuum of speech. Diphone segment assembly is a technique for synthesizing a potentially unlimited variety of continuous utterances under computer control. Investigations of this synthesis technique have progressed until the previous methods of off-line computer processing, and punch card input and storage are no longer capable of dealing with the bulk and variety of data that must be handled in order to examine adequately the synthesis of General American English speech. The new diphone synthesis system to be described consists of several components that are compatible with the IBM System/360. The present combination consists of an IBM 360/40 System, an IBM 2250 Graphic Display Terminal, and a digitally controlled terminal analog speech synthesizer.
Transform Method of Fast Digital Synthesis of Unvoiced Speech Sounds
Authors: Manley Harold J.
Published: October 2005

Digital speech synthesis is often carried out by direct convolution of a sequence of unit samples with a sampled impulse response. In the case of unvoiced sounds, this method can lead to the need for an excessively large number of additions per output sample of the synthesized waveform. This is because of the requirement that the synthesized output be essentially Gaussian and possess the correct spectral energy distribution in order to achieve natural-sounding unvoiced sounds. This paper describes a transform method based upon the Cramer-Kolmogarov representation for a stationary process that reduces the required number of additions per output sample by a factor of at least 10, and which has been found experimentally to produce high-quality synthetic speech. The performance of the method is demonstrated by playing a tape recording of the output of an all-digital vocoder that uses this method of synthesis. [Work supported by Air Force Avionics Laboratory, Wright-Patterson Air Force Base, Ohio.]