bluuebunny/crossref_metadata_2025 huggingface dataset:
Background noise for acoustic emission in a boiling water and a pressurized water nuclear-power reactor
Authors: Ying S. P. , Knight J. E. , Scott C. C.
Published: October 2005

The acoustic noise spectra from 0 to 700 kHz of a boiling water reactor (BWR) have been normalized and compared with the available noise data of a pressurized water reactor (PWR). At frequencies near 500 kHz, which is the interesting frequency range for acoustic emission technology, the hydraulic turbulent and cavitation noises of the PWR were about 0 dB for the selected normalization and the noises of the BWR varied from 0 to 15 dB for the various locations that were monitored. Since the intensity levels of acoustic emission signals are on the order of 0 to 20 dB, the PWR background noise should not completely mask them; however, the BWR noise would obliterate some weak acoustic signals in some locations but allow them to be observed in other locations.
Signal processing to improve speech intelligibility in perceptive deafness
Authors: Villchur Edgar
Published: October 2005

A deaf person with recruitment perceives sound as though listening through a volume expander followed by an attenuator, the expansion ratio and attenuation being typically frequency dependent. (Other perceptive aberrations may also be present, of course.) The subject is often prevented from using enough hearing-aid gain to bring weak consonants into the useful dynamic range of his hearing, because this amount of gain would make lower-frequency, high-amplitude vowels intolerably loud. Such subjects commonly find amplified speech to have poor intelligibility. In a preliminary experiment it is established that recruitment in normal subjects, induced by masking or simulated by expansion of the signal, reduces the intelligibility of amplified speech severely, and that this intelligibility can be largely restored by signal processing. The implication is that recruitment in deaf subjects is a sufficient cause for loss of intelligibility, whether or not there are other causes. In the present experiments, speech is processed by a two-channel amplitude compressor whose frequency-dependent compression ratio is adjusted to compensate the recruitment of the individual subject, and the compressed speech is subjected to frequency-selective amplification similarly adapted to the subject. The aim is to amplify each acoustical element of speech, at each frequency-amplitude coordinate of the speech band, to a relative loudness for the deaf subject corresponding to the relative loudness of that speech element perceived by normals. This processing improved speech recognition, both in quiet and in the presence of competing speech introduced before processing, for six perceptively deaf subjects. Subjects showed an improvement in either initial- or terminal-consonant recognition of at least 22% and as much as 160% at optimum levels in quiet, and from 10% to 229% with speech interference 10 dB below the pre-processed signal.
Energy of the acoustically excited surface wave on a flat semi-infinite elastic medium
Authors: Bunney Robert E. , Goodman Ralph R.
Published: October 2005

An analysis is made of the distribution of intensity of the evanescent waves in a semi-infinite elastic medium when a plane acoustic wave in a contiguous fluid medium is incident upon it. It is found that the maximum intensity and total energy in the evanescent wave occurs at the angle of incidence corresponding to the Rayleigh angle [θR = sin−1(C 0/CR), where CR is the Rayleigh wave speed and C0 is the sound speed in the fluid medium]. This result gives insight both mathematically and physically as to what should be expected for surfaces with curvature as well as for plates.
Origin of the low-frequency sound absorption in sea water
Authors: Yeager Ernest , Fisher F. H. , Miceli John , Bressel Robert
Published: October 2005

Temperature-jump measurements in sea water over the time domain 10−5 to 10−2 sec reveal a single relaxation of 1.5×10−4 sec at 9.7°C, which appears to correspond to the ∼ 1 kHz relaxation found from long-range sound propagation. Laboratory measurements indicate the relaxation to originate from boron with the B(OH)3-B(OH)4− equilibrium the likely relaxation process.
Comments on “Local radiation impedance of vibrating prolate spheroids” [G. C, Lauchle, J. Acoust. Soc. Am. 51, 1106–1109 (1972)]
Authors: Van Buren A. L.
Published: October 2005

The conclusions reached by Lauchle on the local modal radiation impedance of vibrating prolate spheroids are examined in light of the frequency dependence of the surface normal-velocity distributions associated with these modes.
VHF ultrasonic attenuation in mammalian tissue
Authors: Kessler L. W.
Published: October 2005

The ultrasonic attenuation has been determined in mammalian kidney tissue at 100 and 220 MHz. The attenuation, which is known to exhibit a linear dependence between 0.3 and 10 MHz, continues to be linear to about 100 MHz after which a square law (or greater) dependence appears to exist.
Convolution of sonar signals
Authors: Das P. , Blackstock David T.
Published: October 2005

This paper presents the theory of a convolutor which utilizes nonlinear acoustic interaction in fluids.
Comments on animal sonar symposium
Authors: Griffin Donald R.
Published: October 2005

The precision and refinement of echolocation as practiced by bats and cetaceans is proving to be more and more surprising and significant as successive improvements are achieved in the procedures and apparatus used to analyze these biological sonar systems. The multiple and overlapping specializations of animal echolocation are one of their most striking features. Different species of bats employ specialized types of orientation sounds to locate and discriminate among small targets obscured by noise and clutter. Furthermore each bat adapts its sonar technique continuously as it shifts rapidly from one form of echolocation behavior to another. Further discoveries of equal significance will result from improvements that can readily be anticipated in instrumentation for measuring ultrasonic sounds of low intensity in air. Critical observations of orientation behavior under natural conditions will continue to be especially important, because many species do not reveal their full capabilities for complex echolocation under laboratory conditions.
Mechanism of the short-term poststimulatory depression of the cochlear microphonics (hysteresis)
Authors: Legouix J. P. , Pierson A.
Published: October 2005

The characteristics of the short-term poststimulatory depression (hysteresis) have been studied by recording cochlear microphonics (CM) with differential electrodes at several locations along the guinea-pig cochlea. The depression appears, with high-intensity tones, in regions where the vibratory amplitude of the cochlear partition is the largest. It is accompanied by a negative summating potential (SP) whose time course follows exactly the variation of the CM amplitude and in particular outlasts the presentation of the stimulus. At the same time, complex hydrodynamic effects are observed by microscopy. They result in a flow of perilymph from scala vestibuli toward the vestibular spaces, and the membranes of the cochlear partition are displaced in an extreme position. These displacements might provide a protective mechanism against acoustic trauma.
An absorber theory of acoustical radiation
Authors: Shamos Michael Ian , Tavel Morton A.
Published: October 2005

An absorber theory of acoustical radiation is developed by analogy to Wheeler-Feynman electrodynamics. Using a method due to Feynman the acoustic index of refraction for a system of soft spherical scatterers is calculated. The effects of these scatterers on an acoustic source are summed to obtain the usual radiation reaction.
Some ray and intensity solutions in the complex plane
Authors: Krol H. R.
Published: October 2005

This paper presents a method to test range-dependent ray-tracing programs.The method is based on complex-function theory. From an analytic function that has a zero point at the emitter, expressions for sound speed, ray paths, travel time, and intensity can be derived. The intensity expression is very simple and also applicable to caustic points. Two examples are given.
Mouth resonance effects in the flute
Authors: Coltman John W.
Published: October 2005

Resonance of the mouth cavity while playing the flute has been found to occur near 1000 Hz. Experiments with an artificial mouth show the Q of this cavity is less than two when air is passing through the lips. The presence of this coupled cavity can affect the flute frequency by as much at 10 cents, and may increase the losses in the system by as much as one-third.
Backscattering of sound from the ocean volume between Vancouver Island and Hawaii
Authors: Scrimger J. A. , Turner R. G.
Published: October 2005

Integrated scattering-strength measurements describing the backscattering of sound from the ocean volume in winter were obtained along a triangular track running west from the southern tip of Vancouver Island, British Columbia, thence south to Hawaii and then roughly northeast on a return leg to Vancouver Island. Thirty stations were occupied enroute, alternately at midnight and midday, weather permitting. The data were analyzed to yield spectral characteristics of the backscattering from various depth ranges at all station locations. The integrated scattering strength is seen to decrease and the daytime deep scattering layer to become shallower as one proceeds north from Hawaii. Differences between daytime and nighttime scattering and distinctive spectral characteristics associated with sub-tropic and sub-arctic water masses are illustrated.
Transmission-loss measurements using a correlation technique
Authors: Donato R. J.
Published: October 2005

The merits and demerits of using a time-discrimination technique to measure transmission loss through a wall are discussed. The method differs from those previously reported in that a wideband analogue approach is used throughout and the discrimination is improved by the use of a pre-whitening filter before correlating.
A computer simulation of the generation and distribution of cochlear potentials
Authors: Strelioff David
Published: October 2005

A model for the generation and distribution of electrical potentials within the guinea pig cochlea was simulated by computer. Electrical impedances of the cochlear tissues and biological batteries in the stria vascularis and organ of Corti were modelled by a three-dimensional network of resistors and batteries. It was assumed that cochlear-partition movements caused electrical-impedance changes in hair cells and resulted in modulation of steady-state electrical currents normally flowing throughout the cochlea. The model response to a 1000-Hz tone demonstrated that the amplitude envelope for the electrical current through the hair cells had the same shape as the mechanical traveling-wave envelope, whereas the envelopes for the potential changes within cochlear scalae were broader and had their peaks nearer the basal end of the cochlea. Potentials generated by the model accurately reflected the amplitude and phase of the cochlear-partition displacement only from the base to within about 3 mm of the peak of the traveling-wave envelope. An important finding was that, when the impedance of a hair cell was decreased, current from adjacent hair cells was shunted through it, thus providing a possible excitation-inhibition mechanism for neural sharpening.
Effects of anoxia on neural activity in the Xenopus laevis lateral line
Authors: Dodson T. , Strelioff D. , Honrubia V.
Published: October 2005

Spontaneous and mechanically and electrically evoked neural activity in the isolated Xenopus laevis lateral line preparation was studied during changes in the oxygen tension of the bathing solution. Preparations remained excitable and exhibited spontaneous activity (i) for 6–8 hours when the oxygen in the bath was renewed every 10–15 min, (ii) for 1–2 hours when the oxygen was not renewed, or (iii) for 15 min when the oxygen was replaced with nitrogen. Resulting changes in spontaneous and evoked activity were reversible upon readministration of oxygen, but not by addition of glucose to the bath. Since amphibian nerve fibers remain excitable for several hours in oxygen-depleted medium, the experimental results must reflect the effects of anoxia on hair-cell activity. While hair cells have rich cytoplasmic stores of glycogen, it appears that anaerobic metabolism is insufficient for maintaining normal synaptic function. [Supported by grants from the Deafness Research Foundation und USPHS 08193.]
Walk-away test for determining room characteristics
Authors: Moreland J. B.
Published: October 2005

If the distribution of the sound-pressure level around a machine in an ordinary room is explored, it should be possible to deduce many important properties of the source and the room. For example, it may be possible to predict the effect of adding a certain amount of sound absorption to the room. It may also be possible to predict the acoustic behavior of the machine in a room with different acoustic properties. This paper describes a practical method for accomplishing these predictions. Examples are given.
Improving the acoustic environment for in situ machine noise measurements
Authors: Angevine O. L.
Published: October 2005

When using in situ measurements to characterize the noise of a machine in other installations, the measurement space must meet certain qualifications, as outlined by Hübner. In cases where nearby walls and other reflective surfaces interfere, such surfaces may be given a temporary sound-absorptive treatment. If the machine is symmetrical, measurements on an unobstructed side can be accepted as typical of both sides. Temporary barriers may be used to shield measuring microphones from the direct noise of other nearby noise sources which cannot be turned off. While some installations are hopelessly unsuitable for measurements of typical machine noise, such methods can often make an installation acceptable. Similar approaches can be applied to special-purpose facilities used for noise measurements of machines and engines. An effective facility can be developed at minimum cost for making direct field measurements to distances of at least 5 ft. A recently developed facility of this type in a vacant automotive collision shop was made by covering less than 50% of the hard walls and ceiling with panels of acoustic absorption. Nearly spherical spreading (4–6 dB per distance doubling) was obtained out to 24 ft for frequencies above 250 Hz.
Technique for improved accuracy and robustness of linear prediction coefficient estimation
Authors: Allen J. B.
Published: October 2005

A technique is discussed for increasing the accuracy, time resolution, and robustness of the linear prediction method in speech analysis. The technique consists of generating statistics on a large number of coefficient estimates using short frame lengths, and then defining the mode of the distribution as the estimate. Examples from synthetic and real speech are given.
Digital hardware for pitch detection
Authors: Dubnowski J. J. , Schafer R. W.
Published: October 2005

A pitch detection algorithm, designed for robust performance and hardware simplicity, has been studied by simulation and has been implemented in digital hardware. The speech signal is low-pass filtered at about 900 Hz and sampled at a 10-kHz rate. A value of +1 is assigned to a speech sample if that sample value is above a positive threshold; likewise, a value of −1 is assigned if the sample is below a negative threshold. Otherwise, a value of 0 is assigned to the sample. The clipping thresholds are adapted to the speech level within a 30-msec analysis interval. The short-time autocorrelation function of this infinitely center clipped speech signal exhibits a strong peak at the pitch period for voiced speech and no peak for unvoiced speech. The peak location serves as the estimate of pitch period. A similar approach was proposed by Sondhi [IEEE Trans. Audio Electroacoust. (June 1968)]. Since only two data bits per sample are required to represent the infinitely center clipped speech, memory requirements are minimized and the computation of the autocorrelation function is reduced to a simple logical combination. The algorithm is easily implemented with standard TTL logic for real-time operation. The system performs well on bandpass speech (missing fundamental) and under conditions of low signal-to-noise ratio. The hardware was constructed for use in a speaker verification system, but can be easily adapted to any application where pitch detection is required.
Coherence of harmonically related CW signals
Authors: Yen N. , Devilbiss G. , Martin R. L.
Published: October 2005

Results of measurement of the coherence of harmonically related CW signals at long ranges are presented, together with a theoretical analysis. In a nondispersive medium, fluctuation in the relative phase angle caused by linear frequency-dependent parameters are relatively low for harmonically related signals. In an experiment conducted with a moving source 500 NM away from the receiver, the relative phase angle stayed unchanged or varied quite slowly in most cases, except around a fading zone. The theoretical analysis of the effect on the phase angle relationship of receiving array configurations, multipath interference, and signal-to-noise conditions is given to explain the observed phenomena. Based on this study, a method of processing the received signal is described which permits the determination of the relative phase angle of two harmonically related CW signals propagating at a distance from the source.
Interference rejection by amplitude shading of sonar transducer arrays
Authors: Wang H. S. C.
Published: October 2005

It is demonstrated that strong coherent interference incident from arbitrary directions can be rejected by amplitude shading techniques previously reported by the author [H. S. C. Wang, “Amplitude Shading of Sonar Transducer Arrays,” J. Acoust. Soc. Am. 55, S70(A) (1974)]. For a given spatial distribution of interference field, the total interference power received by the array is formulated and superimposed on the power received from a hypothetical isotropic noise field. The total noise plus interference power is then minimized with respect to the shading coefficients and subject to the constraint that the mainbeam peak be maintained at unity. As a test case, we have calculated the amplitude shading coefficients for a linear array with logarithmically spaced transducer elements to reject an interference field uniformly distributed over a 15° × 36° solid angle with an intensity of 30, 50, and 70 dB above the noise field. In each case, the directivity pattern of the array adjusts itself by bunching several null points in the direction of interference incidence to achieve a nearly complete rejection of interference without the use of phase shading. Feasibility of developing a self-adapting beamformer based on this principle is being investigated.
Quantitative distribution of alleged transmitter amino acids in the cochlear nucleus
Authors: Lowry Oliver H. , Godfrey Donald A. , Carter Joyce A. , Matschinsky Franz M.
Published: October 2005

Glycine, GABA, glutamate, and aspartate may serve transmitter functions in the CNS. Measurements of their distribution within the cochlear nucleus of the cat have been made with quantitative micro methods. The tissue is quick frozen, and sectioned at −20°C. The sections are dried at −40° and divided under visual control into samples of 0.04–0.4 mm2 in area. Each sample is weighted (0.2–2 μg) and analyzed. The greatest differences in distribution were found for GABA (a 40-fold range in concentration). Glycine values ranged eightfold, glutamate sixfold, and aspartate fourfold. Both GABA and glycine were especially low in the auditory nerve root and were highest in the superficial layers of the dorsal cochlear nucleus. Here, glycine concentrations were as great as have been found anywhere in the nervous system. The presence of high concentrations of the alleged inhibitory transmitter amino acids in this region appears relevant to the evidence for an inhibitory pathway from the anteroventral cochlear nucleus to the dorsal cochlear nucleus. The distribution patterns for glycine and GABA were somewhat similar but did not coincide completely. This is the first report of high glycine levels in the brain. The analytical principles are described.
Adaptation of quantitative biochemical approaches to the normal and experimental inner ear
Authors: Thalmann R.
Published: October 2005

The anatomical complexity of the cochlea poses numerous obstacles to the quantitative biochemical study of its tissues and fluids. For instance, the preparatory stages of the quantitative histochemical method of Lowry cannot be directly applied to the inner ear, primarily because the tissues are encapsulated by bone, which makes frozen sectioning virtually impossible. However, temporal bones can be freeze-dried in toto with preservation of three-dimensional tissue structure. In the organ of Corti three-dimensional preservation offers the advantage that reasonably homogeneous and comparatively large populations of cell types can be dissected out and analyzed by microanalytical techniques. Analysis of individual cell types is of potential importance for the elucidation of the biochemical counterpart of moderate noise trauma (e.g., asymptotic threshold shift) in which gross cell damage is absent but electron-microscopic changes are detectable in the outer hair cells. A topic of great current interest is the profound effect of salidiuretic drugs upon the endolymphatic potential and energy metabolism of the stria vascularis. Pertinent electrophysiological experiments combined with specially adapted biochemical approaches are presented. Difficulties arising from interactions with cochlear fluids and other potential sources of artifact are discussed.
Analysis of memoryless polynomial nonlinearities
Authors: Molnar Charles E.
Published: October 2005

It can be shown that the output of a polynomial transformation Y(t) = a0 + a1X(t) + A2X2(t) + …, where X(t) = A1 cos(w1t + φ1) + A2 cos(w2t + φ2), may be expressed as Y(t) =  ∑ α1=−∞∞  ∑ α2=−∞∞ C(α1,α2) × exp{jα1(w1t + φ1) + α2(w2t + φ2)}, where C(α1,α2) =  ∑ S1=0∞  ∑ S2=0∞ w!aw(A1/2)|α1|+2S1(A2/2)|α2|+2S2S1!(S1 + |α1|)!S2!(S2 + |α2|)! and w = |α1| + |α2| + 2(S1 + S2). We observe that the amplitudes of distortion products do not generally vary with the input amplitude according to a power law, and that the discrepancy from a power law is likely to be greatest if some of the polynomial coefficients bear negative signs, as they do in saturating-type nonlinearities. A second observation, valid if ω1 and ω2 are integer multiples of a common fundamental ω0, is that distortion product amplitudes are also a function of the primary phases angles φ1 and φ2. If the nonlinearity is an exponential function Y(t) = expX(t), it can be shown that C(α1,α2) = Iα1(A1) ×Iα2(A2), where Iu(A) is the modified Bessel function of order u. An extension of these results to the case of three sinusoidal inputs can be used to derive conditions for cancellation-tone experiments.
Characteristics of the (f2 − f1) component in response patterns of single cochlear nerve fibers
Authors: Pfeiffer Russell R. , Molnar Charles E.
Published: October 2005

At a fixed stimulus level, single cochlear nerve fibers are responsive to sinusoidal stimuli over a band of stimulus frequencies. Stimulation with the sum of two sinusoids, each within the response band, can cause (f2 − f1) and (f2 + f1) distortion products (as well as many others) to appear in period histograms with amplitudes and phase angles that are predictable from the phase versus frequency plot for single sinusoids by a memoryless polynomial nonlinearity model. On the other hand, stimulation with the sum of two sinusoids, each above the response band, can for some choices of f1 and f2 produce large (f2 − f1) distortion products in the period histogram without significant response at any other frequency including f1, f2, and (f2 +f1). In this case, phase-versus-frequency plots for the (f2 −f1) component are similar to those for single sinusoidal stimuli at a frequency of (f2 −f1). Also, the amplitude of the (f2 −f1) component exhibits tuning curves that lie within the tuning curve for single sinusoidal stimuli but are shifted toward lower frequency. These observations are consistent with the hypothesis of an (f2 −f2) traveling wave, and are akin to a similar observation for the (2f1 − f2) distortion product reported by others. [This work has been supported in part by grants from the United States Public Health Service.]
Response of a neuron model to sinusoidal stimulation
Authors: Matthews John W. , Molnar Charles E.
Published: October 2005

The response characteristics of a Markov process model for neuron activity [Frederick C. Moses et al., Biophys. Soc. Abstracts 9, 115 (A) (1969)] are investigated. The input to the model is the intensity function of a Poisson process and the output is the intensity function for the firing of the neuron model. The model and its implementation are described by Clark and Molnar in Computers in Biomedical Research [Stacy and Waxman, Eds., (1974), Vol. IV]. In the current study, the input intensity function is modulated with sinusoids and, after the process reaches steady state, the output intensity function is plotted versus time for one period of the stimulus. In the case of a real neuron, a period histogram would be an estimate of this intensity function. Fourier analysis of the intensity function waveforms indicates that the model has considerable frequency selectivity. This frequency selectivity appears to be related to the tendency of the model to oscillate when there is a step in the input intensity function. This model has been previously applied to the study of cochlear nucleus neurons; it is less apparent how model parameters might be related to cochlear nerve fibers. [This work has been supported in part by grants from the United States Public Health Service.]
Monitoring community noise from a tethered balloon
Authors: Chanaud Robert C. , Simmons Robert A.
Published: October 2005

Since August 1973, experiments have been conducted to demonstrate the technical feasibility of monitoring community noise from a tethered balloon. The primary emphasis has been on the development of a simple lightweight aerial package, a commercially available and stable balloon, and a simple ground station. The aerial package used was a GR 1565B, with a lightweight battery pack, and an FM telemetry transmitter operating at 400 MHz. The balloon was an aerodynamically shaped, helium-filled blimp of 114.9-ft3 capacity with a total payload of nearly 4 lb. The ground station had a simple, high-gain antenna and a meteorological-band receiver which delivered a signal to several standard acoustical instruments. Some of the experimental problems in using this equipment are described. The advantages of aerial monitoring for obtaining a data base of community noise levels in a cost effective manner will be discussed. Data obtained from known sound sources and from general community noise are shown and compared with simultaneous ground data.
Important aspects of chain saw noise-measurement procedure and control
Authors: Mutyala Brahmaji , Soedel Werner
Published: October 2005

The purpose of this paper is to discuss noise-measurement and reduction procedures peculiar to chain saws. Thus, outdoor and indoor measurement considerations are discussed extensively. Repeatable measurements are important for engineering development, but general industry standards are too vague to satisfy this requirement. Neither SAE proposed new engine sound-measurement procedure nor Power Saw Manufacturer Association (PSMA) standard sound-measurement procedure mentions anywhere the location of the sound source (height of the sound source from the surface), which is a very important factor in calculating the correction factor due to ground reflections. When the surface is hard, like concrete, the correction factor (total SPL, dB re free field) can be readily computed, but over dirt or grassy surfaces (PSMA standard sound-measurement procedure) rather puzzling results are often obtained, since not only is reflection incomplete, but there is a phase delay δ in the reflected wave. This paper mainly concerns an outdoor sound-measurement procedure peculiar to chain saws. Noise identification procedures are also presented and selected results of a particular case study are shown and discussed.
Parsing and word matching in the Lincoln Laboratory, speech recognition system
Authors: Hall D. E. , Forgie J. W.
Published: October 2005

The Lincoln system is designed to recognize sentences conforming to a context-free grammar. Parsing is guided by a heuristic evaluation function which combines individual word scores into parse path scores. The word scores measure the degree of correspondence between phonemic dictionary spellings and the results of an acoustic-phonetic analysis of the input sentence. Word scoring is based on two computer-generated scoring matrices derived from confusion statistics gathered from 113 sentences. The synchronization problem resulting from missing and spurious segments is simplified by first aligning the vowels (sometimes in more than one combination). The effect of phonological rules is handled by insetting optional phonemes in the dictionary and flagging others as possibly missing. Other rules, which are specific to Lincoln's front-end analysis, predict spurious segments and the effect of neighboring semivowels on vowel classification. [This work was sponsored by the Advanced Research Projects Agency of the Department of Defense.]
Digital hardware for PCM to ADPCM conversion
Authors: Bates S. L. , Schafer R. W.
Published: October 2005

When several different digital coding schemes are used in a speech communications system, there arises the problem of converting with minimum distortion between two digital representations. Direct digital processing is very desirable for this purpose. As an experiment in digital code conversion, a system for converting from linear PCM to adaptive differential PCM has been constructed. In this system, speech is converted to a 12-bit linear PCM representation by a standard A/D converter, and an all-digital processor converts the 12-bit samples to either a three- or four-bit ADPCM representation as described by Cummiskey, Jayant, and Flanagan [Bell Syst. Tech. J. (Sept. 1973)]. The digital processor, constructed out of standard TTL logic, involves a simple arithmetic unit and a library of step sizes stored in read-only memory. Time division multiplexing is used to achieve efficient use of arithmetic hardware. Internal parameters and input sampling rate are easily changed to study their effects on the code conversion process. In addition to demonstrating the principles of digital code conversion, the PCM to ADPCM converter can be used for bit-rate reduction for speech transmission and digital storage. Speech quality is illustrated with recorded demonstrations.
Asymptotic high-frequency behavior of guided infrasonic modes in the atmosphere
Authors: Kinney Wayne A.
Published: October 2005

Refinement of previous theoretical formulations and numerical computations of pressure waveforms as applied to atmospheric traveling infrasonic waves could include a description of their asymptotic behavior at high frequencies. In the present paper, calculations based on the W. K. B.J. approximation and similar to those introduced by Haskell [J. Appl. Phys. 22, 157–167 (1951)] are performed to describe the asymptotic behavior of infrasonic guided modes as generated by a nuclear explosion in the atmosphere. The results of these calculations are then matched onto numerical solutions which have been given by Harkrider, Pierce and Posey, and others. It is demonstrated that the use of these asymptotic formulas in conjunction with a computer program which synthesizes infrasonic pressure waveforms has enabled the elimination of problems associated with high-frequency truncation of numerical integration over frequency. In this way, small spurious high-frequency oscillations in the computer solutions have been avoided. [Work sponsored by Air Force Cambridge Research Laboratory.]
An impedance measuring facility for the low-frequency evaluation of acoustic materials
Authors: Ho L. T. , Heiner L. F. , Shrader J. M.
Published: October 2005

A simple, compact facility has been constructed which measures the specific acoustic impedance directly. It is capable of evaluating the acoustic characteristics of materials for underwater applications as a continuous function of frequency from 100 Hz to 5 kHz. The facility consists of a short, thick-walled tube, a dual-sensor hydrophone, and its associated electronic instrumentation. The tube is a liquid-rifled standpipe with a sound source mounted at the top and a sample material at the bottom. The sample is terminated by a rigid boundary. The dual-sensor hydrophone is positioned inside the tube and its two outputs are processed to yield the acoustic impedance at a point in the fluid column. Acoustic characteristics of the sample are then obtained from these impedance data by a computer. Theories underlying the impedance measuring technique are presented, and the procedure for calculating the acoustic properties from the impedance data is described. Sound velocities of several samples evaluated agreed reasonably well with those obtained by the pulse technique.
Orthogonal excitation of magnetostrictive transducers
Authors: Huston E. Lee
Published: October 2005

A piston-type magnetostrictive transducer consisting of a scroll-wound tube 12 in. long, 1.25 in. in diameter, and of 0.125-in. wall thickness was constructed from 0.008-in. Nickel 200 sheet and fitted with toroid-wound and solenoid-wound drive coils. This prototype device was evaluated using both conventional and orthogonal modes of excitation. Electrical impedance measurements at low drive levels showed an increase in the electromechanical coupling coefficient with orthogonal excitation. Power tests were conducted with output power levels up to 50 W/cm2 of active core area by driving the piston into a tuned ceramic dummy load. Device efficiency in excess of 50% was obtained at 5 W/cm2 output power. As the output power was increased, orthogonal excitation consistently provided higher efficiency and linearity than conventional excitation. Device linearity improved with increasing drive level for the orthogonal tests. Self-biased excitation was obtained by driving the two coils plus a pair of diodes in parallel. For self-biased excitation, device efficiency increased with drive level exceeding 30% at 40 W/cm2 output power. These results confirm the earlier theoretical predictions [J. Acoust. Soc. Am. 55, No. 5 (1974)] for orthogonal excitation.
Perception of suprasegmental and segmental features
Authors: Garnes Sara
Published: October 2005

This study reports the results of an experiment in which perception of a suprasegmental feature dominated perception of a segmental feature. The test language is Icelandic. Stimuli based on a minimal triple /ka:ka/ “cake,” /kak:a/ “keg (oblique),” and /kahka/ “to heap up,” were prepared on a terminal analog synthesizer. Variables include duration of V1, C2, and the absence or presence of four durations of preaspiration, [h]. If no preaspiration is present, subjects choose /kak:a/ or /ka:ka/, depending on the ratio of V1/C2 durations. If preaspiration is present, subjects choose /kahka/ or /ka:ka/, again only with favorable V1/C2 ratios. The fact that the word /ka:ka/ is chosen, In spite of the presence of an otherwise misplaced segmental feature indicates that perception of the suprasegmental feature length dominates perception of the segmental feature preaspiration. Evidence from studies of production and perception of American English is discussed which supports the observation made here. A model of perception in which suprasegmental feature detection is warranted. Such a model—in the form of a decision tree—is proposed.
Proposed “vehicle noise number” requirement for new motor vehicles
Authors: Hellweg Robert D. , Nechvatal Michael F. , Schomer Paul D.
Published: October 2005

Noise-control regulations for new motor vehicles should be based on the operating modes of these vehicles in the community. The proposed Illinois noise-control regulations for new motor vehicles include performance standards based upon the operating modes for each of five classes of vehicles: heavy trucks (GVW: over 24 000 lb); buses and medium trucks (GVW: 14 000–24 000 lb); light-medium trucks (GVW: 8000–14 000 lb); cars and light trucks (GVW: under 8000 lb); and motorcycles. One performance standard is the “vehicle noise number,” which is a composite rating dependent upon maximum sound level, medium-speed cruise sound level, and “urban acceleration” sound level. The “urban acceleration” sound level is defined as the peak A-weighted sound level measured 50 ft from the vehicle path, when a vehicle traverses 100 ft from rest at a quasiconstant throttle sorting in a specified traversal time. The specified traversal time for each class of vehicles was obtained from an in-use survey of the time to traverse 100 ft for vehicles accelerating from rest. “Urban acceleration” sound levels for automobiles, which were obtained under test conditions, are discussed.
Study of the repeatability of noise measurements at motor vehicle test sites
Authors: Hemdal John F.
Published: October 2005

Variations in the sound-pressure level occur during vehicle passby noise measurement tests (as in SAE J986a). Only a portion of these variations are a result of changes in the noise power or radiation characteristics of the vehicle. Other contributions to measurement variation have been referred to as site factors and include the effects of site geometry, ground impedance, meteorological conditions, and variations in micrometeorology. This latter is termed turbulence in this study and refers to small-scale changes in temperature gradients and wind gradients. This study was an effort to determine the contribution to the variation by turbulence and changes in site geometry. A standard sound source was placed 15 m from a calibrated microphone. The sound-pressure level determined from the microphone was monitored during several 1-min tests and the variation of sound-pressure level was compared to meteorological conditions. Effects of site geometry were estimated by alternately substituting source and microphone at the same site. This paper describes the test details, compares measurement variation with meteorological conditions, and discusses the manner in which variations in the principal factors affect the measurement results. [Research supported in part by Ford Motor Company.]
Taxiing noise of jet aircraft
Authors: Hershey Robert L. , Turner Louie
Published: October 2005

Taxiing noise measurements at Atlanta Airport included 404 measurements of maximum passby sound level at a distance of 50 ft from the edge of the taxiway. The following average sound levels were measured: Boeing 727—98.6 dBA, Boeing 737—100.1 dBA, Convair 880—99.8 dBA, DC-8—102.5 dBA, DC-9—98.8 dBA, and DC-10—93.8 dBA. The primary purpose of these measurements was to determine whether changes in noise level result from shutting off some engines during taxiing for fuel conservation and air pollution reduction. Measurements of taxiway passby and stationary engine runups indicated that engine shutoff procedures reduce passby levels by 5 dBA for the DC-8 and 2 dBA for the 727. The reduction is 10 dBA for the DC-9 when measured on the side with the engine shut off. Measurements were also made of the ambient noise inside the terminal building and atop the terminal roof. After analysis of the noise data from this test series, it was concluded that although the noise levels of individual planes on the taxiways were reduced by engine shutoff, the noise environment of the air terminal building was unaffected, because of the large distances from taxiways involved. [Work supported by Federal Aviation Administration.]
Comparison of airport measurements of approach noise produced by jet aircraft
Authors: Bishop Dwight E.
Published: October 2005

In 1971, field noise measurements were made during two 10-day periods at locations under the ILS approach path to Runway 21R at Detroit Metropolitan Airport to evaluate proposed changes in glide slope intercept altitudes [FAA-RD-71-117 (Dec. 1971)]. Over 1400 recorded flyovers were analyzed to obtain effective perceived noise level (EPNL) values and other noise measures. Regression lines correlating noise levels with slant distance over observed distances to aircraft ranging from about 1500 to 6000 ft show good agreement with published noise curves frequently used for noise exposure forecast (NEF) calculations. There is relatively large variability in data, however; 95% confidence intervals about the regression lines ranged from approximately 7 to 9.5 dB. Also examined is the observed variability in noise levels at a single ground position under the approach path. Standard deviations of the order of 5 to 8 dB were noted. Of the several sources of variability in noise levels at this position, distance variations appeared to be the most important.
Level dependence in the two-tone masking experiment
Authors: Bilger R. C. , Wolf R. V.
Published: October 2005

The two-tone masking experiment used in the direct estimation of critical bandwidth was conducted at two sound-pressure levels of the masking tones, 50 and 90 dB SPL, on three adult listeners with normal hearing and two with moderate (45 and 50 dB THLs), flat, unilateral, sensorineural hearing losses. The test signal was a narrow band of noise (±80 Hz) centered around 1000 Hz. The two-tone maskers were 1000 Hz±40, ±60, ±80, ±100, ±150, ±200, ±300, and ±500 Hz. In addition to the two-tone masking experiment, control experiments in which each of the single tones of each two-tone pair were used as the masker were conducted at both levels. For listeners with impaired hearing, the experiments were run at both levels in their normal ear but only at the 90-dB level in their impaired ear. For the 90-dB masking level, the two-tone masking experiment failed to show the sudden diminution in masking that defines the critical band at the 50-dB level. This failure to observe the CBW at the 90-dB level can be explained in terms of the control experiments, which indicated that masking by the low tone alone remains relatively constant across frequency for a 90-dB level of the masker.
Masking beats with beats
Authors: Taylor M. M. , Brenneman H. A. , Smith S. M.
Published: October 2005

The envelope of the output of any filter whose input is more complex than a sinusoid may be described in terms of the spectrum of beats among the frequencies in its input. Taylor and Smith [J. Acoust. Soc. Am. 53, 377(A) (1973)] suggested that one beat might mask another. We now propose that masking of beats by beats is analogous to masking of tones by tones, and that the auditory system has beat-sensitive filters whose inputs are the output envelopes of the primary (critical-band) filters. In the experiments, the frequencies of two masker tones were held constant. As a function of signal frequency, masking was greatest when the beat between the signal and the nearest masker had the same frequency as that between the two maskers. For at least some listeners, the amount of masking depended primarily on the intensity of the softer, not the louder masker. For maskers at 470 and 490 Hz, the half-bandwidths of the masking maxima at 450 and 510 Hz were on the order of 5 Hz, which represents a rough estimate of the half-width of the “20-Hz beat at 500-Hz” filter.
Detection threshold for a two-tone complex
Authors: Sondhi Man Mohan , Hall J. L.
Published: October 2005

This paper reports on a measurement of the absolute threshold for a two-tone signal as a function of the frequency separation between the tones. When interpreted in terms of an energy detection model (peripheral filter followed by squarer, integrator, and threshold detector), the measurements give an estimate of the time constant of the integrator. This time constant has been estimated by many investigators, and published estimates range from 10 to 500 msec. Our measurements on two subjects give a value in the neighborhood of 150 msec.
Fundamental nonlinear equations of atmospheric acoustics: a synthesis of current physical models
Authors: Pierce Allan D.
Published: October 2005

Paper represents an attempt to give concise but comprehensive set of macroscopic continuum equations incorporating all physical mechanisms of significance in atmospheric acoustics. While recent literature on sound absorption [e.g., Evans, Bass, and Sutherland (1972); Calvert, Coffman, and Querfeld (1966)] implicitly contains such information, discussion tends to be centered on the particular (albeit fundamental) case of plane waves of infinitesimal amplitude and constant frequency in a homogeneous quiescent atmosphere. Two problems of current interest (attenuation over distances of tens of thousands of kilometers of ducted acoustic gravity waves and the nonturbulence contribution to rise times of sonic booms) cannot be discussed in this context. The resultant set of equations incorporates Swartz, Slawsky, Herzfeld, Tanvzos, Shields, Evans, Bass, Sutherland, and Piercy models in which different temperatures are assigned to different internal vibration modes. Stokes's (1951) original model of electromagnetic heat radiation transfer is incorporated as a somewhat regretful compromise since Smith's (1957) model is restricted to plane waves. Value of q is estimated from recent work by Calvert et al. [Work sponsored by Air Force Cambridge Research Laboratory.]
Cognitive factors in masked frequency discrimination
Authors: Hafter E. R. , Carrier S. C.
Published: October 2005

Frequency discrimination in noise, using signals lying either within a critical band or within different critical bands, presents a paradigm to examine the question of serial versus parallel processing of incoming sensory stimuli. Three observers, two musically naive and one musically trained, were tested for frequency discrimination using a two-interval forced-choice procedure. Equally detectable 100-msec tonal signals, masked by continuous broad-band noise, appeared in each of the two intervals. One tone was always 500 Hz while the other was 500 Hz plus Δf, and observers were required to indicate which interval contained the higher frequency. Eight Δf's were used, ranging from 6.3 to 1636 Hz. For the musically naive observers, the percentage of correct responses [P(c)] improved with increasing Δf until a plateau was reached at a Δf of 25 Hz. This was true for both moderately [P(c) = 70%] and highly [P(c) = 90%] detectable signals. In both cases, asymptotic performance was at a percent correct corresponding to the detectability of either tone alone. This result suggests serial processing, since the listeners appeared unable to monitor simultaneously the output of two critical bands during the duration of the signal. The musically trained observed also showed asymptotic performance, but at a P(c) higher than that for detection alone. This latter result suggests a parallel process. These data differ from those of a previous study by Tanner [J. Acoust. Soc. Am. 28, 882–888 (1956)], where all subjects could be said to be serial processors, and as such this may reflect differences between the ways in which musicians and nonmusicians process frequencies. [Supported in part by grants from the National Institutes of Health and from Oberlin College.]