bluuebunny/crossref_metadata_2025 huggingface dataset:
Effect of combinations of sodium salicylate and noise on the auditory thresholds of the chinchilla
Authors: Woodford C. M. , Henderson D. , Hamernik R. P.
Published: October 2005

Thirty-nine monaural chinchillas were used to study the interaction between sodium salicylate and various TTS-producing noise paradigms. Five animals were included in each of the following three groups: (a) sodium salicylate (400 mg/kg) plus 2–4 kHz, 95-dB noise band for 1 hour; (b) sodium salicylate (400 mg/kg) plus 4-kHz CF noise band at 80 dB for 96 hours; and (c) sodium salicylate (400 mg/kg) plus 50 noise impulses having 50-μsec duration on 153-dB peak SPL, presented 1/min. The remaining 24 animals served as various controls in groups exposed to sodium salicylate or the noise paradigm alone. Thresholds were measured before, during, and after exposure using the AER technique. When sodium salicylate was combined with the various noise conditions, the maximum TTS values obtained from the combination studies were comparable to those obtained from the single agent producing the greatest TTS at a particular frequency. No consistent alteration in either magnitude or time course of post-treatment threshold shift was found following the combination treatments as compared to the individual agents alone. Hazard to the auditory system resulting from a combination of sodium salicylate and noise was concluded to be no greater than the hazard presented by either agent alone.
Changes in scala media potentials produced by ethacrynic acid
Authors: Honrubia V. , Strelioff D. , Sitko S.
Published: October 2005

Following intra-arterial administration of ethacrynic acid (EA) in the guinea pig (50 mg/kg), the endocochlear potential declines from its normal value of + 80 mV to − 30 to − 40 mV, similar to that during asphyxic anoxia. The regression line describing the change in CM amplitude as a function of EP amplitude is statistically identical for both anoxia and EA administration, with a slope of 0.49 and a correlation coefficient of 0.96. Perfusion of scala vestibuli with isotonic KCl had no significant effect on the value of the negative EP resulting from EA administration. These results suggest that (i) changes in CM amplitude following EA administration are due solely to the depression of the EP resulting from the action of EA on energy production in the stria vascularis, and (ii) the negative EP is not the result of a K+-diffusion gradient across Reissner's membrane. [Supported by grants from the Deafness Research Foundation and USPHS 08193.]
Noise-induced reduction and recovery of chinchilla cochlear microphonic response
Authors: Drescher D. G.
Published: October 2005

Anesthetized chinchillas were maintained in quiet for 24 to 48 hours with virtually no loss of maximum CM voltage or of CM sensitivity. Alteration of body temperature (and parallel change in cochlear temperature) from 29° to 39°C had little effect on normal CM response. With presentation of steady octave-band noise centered at 1 kHz at 90 dB SPL overall, CM progressively decreased and approached an asymptote at a rate dependent on body temperature; the rate of reduction was less at lower temperatures and greater at higher temperatures. In addition, CM tended to remain near initial values longer at lower temperatures. With termination of the noise, at higher temperatures there was an initial rapid recovery of CM followed by a slower recovery toward normal CM values for the duration of the experiment. At lower temperatures, recovery of CM appeared to be retarded. These studies are interpreted as indicating that noise-induced reduction of cochlear response is linked to metabolic processes. [Supported by NINDS Grant NS 03856.]
Sound power measurements on large machinery installed indoors—two-surface method
Authors: Diehl George M.
Published: October 2005

Sound power ratings of machinery are becoming more important for a number of reasons. Industrial installations must comply with recently enacted state and local noise control codes, and sound power ratings are needed to predict compliance. In addition to this, the Federal Noise Control Act of 1972 requires labeling of machinery to show maximum noise emission, and this, logically, may be in terms of sound power. There are several ways to determine the sound power of machinery under laboratory conditions, but these procedures are usually not applicable in industrial environments. The two-surface method offers the best practical approach to the problem of calculating sound power ratings of large machinery installed indoors, under actual operating conditions.
Qualification procedures for free-field conditions for sound power determination of sound sources
Authors: Hübner G.
Published: October 2005

For determination of sound power of sources by the “method of enveloping measurement surfaces,” the test environment should provide a measurement surface which lies (1) outside the nearfield of the sound source under test and (2) inside a sound field free of undesired sound reflections from room boundaries or reflecting objects near the source. Methods to check the free-field conditions and to qualify a given measurement surface for an actual source under test are (1) the absolute comparison test using a (small) calibrated reference sound source, (2) the relative comparison test using a small test sound source which radiates broad-band noise that remains essentially constant during the measurement, and (3) the reverberant test, which requires measurement of reverberation time. Method 3 is only applicable in closed spaces (rooms). Methods 1 and 2 may be used in rooms and outdoors. Methods 1 and 2 require replacing the source under test by the reference sound source or test sound source in the test site. If the source under test cannot be removed, methods 1 and 2 still allow qualification for free-field conditions, with less accuracy. This paper deals mainly with the relative comparison test (method 2) and gives information about the accuracy of the determination of the environmental corrections factor K under different field conditions.
Investigation of procedure for free-field measurement of sound power above a reflecting plane
Authors: Holmer Curtis I.
Published: October 2005

This paper provides a preliminary report of an experimental investigation of the accuracy and precision of various measurement methodologies for determining sound power output of “large” machines in the free field over a reflecting plane out of doors. The purpose of this investigation is to place error bounds, having some statistic validity, on many of the free-field measurement procedures currently proposed or in use and, in particular, to compare the results of “nearfield” and “farfield” measurements. The sources used for the investigation included 17 portable air compressors of various type (powered by internal combustion engines), a “reference” sound source, and a loudspeaker driven by a pure-tone source. The data recorded include sound-pressure level (A-weighted, linear, and 1/3-octave band) on an 84-point hemispherical array of 7-m radius, and “nearfield” measurements, sampled every square meter, on a rectangular surface one meter from the machine surface. In addition, normalized variance of RC integrated mean-square pressure in 1/3-octave bands was determined at each position for use in assessing signal variability and spectral content. These data are being reduced to provide information on the deviation (using the 84-point farfield data as a reference) of sound power measurements, using subsets of the data as appropriate to various methodologies. In this presentation we discuss the measurement procedures in detail, review some selected results of the data analysis, and discuss some of the major sources of measurement error. [Work supported in part by EPA/ONAC.]
Use of the correlation between surface velocity and nearfield pressure for measuring the noise of a large machine in situ
Authors: Hodgson Thomas H.
Published: October 2005

It is well known that one method of sound power accounting of the noise sources present in a large machine in situ is to measure the nearfield mean-square pressures over the surface of the machine. However, this procedure is likely to be in error if either some areas of the machine are in such close proximity that certain nearfield pressure measurements are contaminated by radiation from other parts of the machine or if the room conditions are very reverberant. An attempt is described to rank order the sound power output of the areas of a large machine using measurements of the correlation coefficient between the surface velocity and the nearfield at points over the machine.
Noise measurement method for large transformers
Authors: Fader Bruce
Published: October 2005

Existing methods of reporting noise levels caused by large transformers have shortcomings because of the directional radiation usually encountered with these machines. A new method of measurement and evaluation for transformers has been developed. Several measurements have been made by this method on substation transformers of from 7.5 to 90 MVA. Radiation patterns are clearly indicated and the distribution of sound energy both by direction and octave band is shown. Strengths and weaknesses of the new method are discussed.
Improved measurement of turbine-generator sound levels in semi-reverberant spaces with directional microphones
Authors: Niskode P. M. , Hribar A. E.
Published: October 2005

The sound field in a turbine-generator hall approximates that of a reverberant field. Accurate measurements of the direct machine sound levels cannot be made except very close to the machines (i.e., in the nearfield) because of the reverberant sound and other equipment noises. The potential of directional microphones to reduce the influence of these noises on the sound-level measurements of turbine-generators was examined from a theoretical and experimental point of view. In comparison with an omnidirectional microphone, the theory shows that a directional microphone essentially “extends” the direct field of a source in a reverberant environment to greater distances. Thus, at a given location from a machine, a more accurate measurement can be made with a directional than with an omnidirectional microphone. The magnitude of the improvement is a function of the source directivity, the microphone directivity, the turbine hall room absorption and the distance from the source. Because of its directivity, a directional microphone could also be used to improve measurements of the relative noise contributions of several closely spaced sources. Comparative field measurements in turbine-generator halls with omnidirectional and directional microphones are presented. Comparisons between the theory and the measurements are presented and the resulting implications are discussed.
Twenty-one years in speech synthesis
Authors: Lawrence Walter
Published: October 2005

The history of speech synthesis is reviewed, with particular emphasis on the purpose for which each of the many speech synthesizers was designed. During the two decades reviewed, there has been progress both in the instrumentation and in the nature of the problems that speech synthesis has been used to elucidate. The paper is illustrated by recordings of early synthetic speech, and a live demonstration of the latest number of the PAT family of parametric artifical talking machines.
Optimal number of poles in a linear prediction model
Authors: Makhoul John , Cook Craig
Published: October 2005

The problem of choosing an “optimal” number of poles in fitting an all-pole model can be formulated as an estimation problem with an associated measure of fit. One measure of fit is the information theoretic criterion proposed by Akaike as an estimate of the mean log likelihood. Under the assumption that the signal is Gaussian, the criterion can be given as I(p) = logVp + 2p/N, where Vp is the normalized linear prediction error, p is the number of poles, and N is the “effective length” of the signal I(p) is computed as a function of p up to the maximum value of p that is of interest. That value of p for which I(p) is minimum is taken as optimal and the resulting model is said to characterize the given signal adequately. This criterion can be used in a speech compression system to determine an adequate predictor order for each frame of the analysis. Only that number of parameters need be transmitted, resulting in a lower transmission rate. Another possible application is in speech recognition, where the “optimal” predictor order can be used as a measure of formant structure.
Towards a minimally redundant linear predictive vocoder
Authors: Viswanathan R. , Makhoul John
Published: October 2005

We consider pitch-excited linear predictive vocoders, and present approaches to achieve minimal transmission rates while maintaining good-quality speech. Our contributions deal with quantization and encoding of predictor parameters before transmission, and their interpolation before synthesis. For purposes of quantization and transmission we have found the so-called log area ratios to be optimal in that they preserve filter stability under quantization, and they possess uniform spectral sensitivity characteristics (Makhoul and Viswanathan, BBN Report No. 2800). The quantized parameters are encoded using (i) a variable-rate scheme that transmits parameters only when speech characteristics have sufficiently changed, and (ii) a variable-wordlength scheme (Huffman code) that uses parameter statistics to code parameters with variable numbers of bits. At the receiver, the synthesizer parameters are reset at the corresponding times at which they were extracted during the analysis. In addition, between every two settings, the parameters are further updated pitch asynchronously by periodic interpolation between the settings. With these features, transmission of good-quality 10-kHz sampled speech is achieved at average bit rates as low as 1650 bits/sec.
Variable-wordlength encoding of speech parameters in a linear predictive vocoder
Authors: Cosell Lynn , Makhoul John
Published: October 2005

Huffman coding, a technique which produces optimal variable-wordlength code, can be used to reduce significantly the transmission rates for speech parameters in a linear predictive vocoder, without causing any change in speech quality. Huffman coding is a statistical procedure which minimizes the average transmission rate for a given set of data by coding the most probable values with fewer bits. When tested with a data base generated by eight speakers, it produced an average transmission rate reduction of approximately 20%. Huffman coding can be combined with other types of encoding which improve the probability distributions; it produces efficient code even when the number of values to be coded is not an integer power of two, and, because it is optimal, it can be used as a standard against which to measure other types of encoding.
Stochastic approximation versus adaptive predictive coding
Authors: Goldberg A. J.
Published: October 2005

While offering a considerable simplification in hardware over adaptive predictive coders, stochastic approximation methods must not further degrade voice naturalness if they are to be practical for transmitting digitized speech below 10 kbps. This paper compares the expected performance of these two coding techniques using S/N ratios and subjective voice-quality judgments. After discussing their relative performance, the talk provides audio tape recordings illustrating the speech quality of stochastic approximation and adaptive predictive coders.
Comparison of the intelligibility of predictor coefficient and formant coded speech
Authors: Keeler Lynn O. , Strong William J. , Palmer E. Paul
Published: October 2005

This study describes the analysis and synthesis of speech using linear prediction and formant coding methods. A detailed comparison is made of speech synthesized using 12 predictor coefficients, six predictor coefficients, five formant frequencies and amplitudes, three formant frequencies and amplitudes, three formant frequencies and amplitudes with monotone pitch, and three formant frequencies with monotone pitch and amplitudes calculated by formula. The synthetic speech and the original speech are examined using Griffith's articulation test for rhyming minimal contrasts. Intelligibility scores are measured for each method of synthesizing speech, and confusion matrices are constructed for each method. Possible explanations for the observed confusions are investigated with the view in mind of suggesting improvements to speech analysis-synthesis systems.
Evaluation of synthetic speech
Authors: Dukes K. D. , Olive J. P.
Published: October 2005

Two paragraphs of computer-generated speech, with the same text, one made at Bell Laboratories by concatenating words spoken in isolation and the other at Haskins Laboratories using a phonetic dictionary, were evaluated with respect to the ease with which listeners could comprehend and retain information about the paragraph. A paragraph with the same text was recorded by a human speaker and used as a reference for comparison. Different groups of subjects listened to one of these three versions, hearing it one, two, or three times, and then answered questions concerning the content of the paragraph. The results indicate that for all three types of speech, comprehension measured in this way increased as a function of the number of times the paragraph was heard. The data indicated that there is a significant comprehensible difference between the two synthetic speech types. Scores for the speech generated by concatenating words spoken in isolation were closer to those for natural speech than were the scores obtained for speech synthesized from a phonetic dictionary.
Comparison of three methods of extracting resonance information from predictor coefficient coded speech
Authors: Christensen Randall L. , Strong William J. , Palmer E. Paul
Published: October 2005

Three methods of extracting resonance information for speech from predictor coefficients are compared. The methods are finding roots of the polynomial in the denominator of the transfer function using Newton iteration, picking peaks in the spectrum of the transfer function, and picking peaks in the negative of the second derivative of the spectrum. A relationship was found between the bandwidth of a resonance and the magnitude of the second derivative peak. Data, accumulated from a total of about two minutes of running speech from both female and male talkers, are presented illustrating the relative effectiveness of each method in locating resonances. The second-derivative method was shown to locate about 98% of the significant resonances while the simple peak-picking method located about 85%.
Study of the use of linear predictor and related methods of speech analysis for measuring vocal-tract area functions
Authors: Engebretson A. M. , Vemula N. R.
Published: October 2005

The linear predictor (LP) and partial correlation (PARCOR) methods of speech analysis have been applied to sounds produced by acoustic horns with inside cross-sectional shapes comparable to the human vocal tract. Our experience with human speech suggests that the area functions obtained from both methods are variable and depend strongly on the type of speech sounds, the sampling rate, the method of windowing and compensation of the signal, and other factors. To evaluate these methods in a controlled situation, the horns are driven with a known input volume velocity. The output volume velocity is measured, sampled at a 20-kHz rate, and analyzed with a computer program using either an LP of PARCOR algorithm. The computed area function and transfer function are compared with the known geometry and measured frequency response of the horn. Implications of these results on the use of LP and related methods for studying vocal-tract area functions in hearing and nonhearing subjects are discussed.
Implementation of an adaptive procedure for linear prediction
Authors: Allen J. B. , Berkley D. A. , Curtis T. H.
Published: October 2005

A gradient algorithm is defined which adaptively minimizes the energy output of a finite impulse response digital filter, for an arbitrary input, by varying the filter coefficients. For a stationary input signal this is equivalent to solving the linear estimation equations for an all-pole system model. The algorithm is simple and is probably easier to implement than the matrix equation methods required by direct solutions of the linear equations. Moreover, real-time hardware using this general approach is already available in data transmission equalizers and echo cancellers. We describe a realization of the algorithm using a small computer and real-time digital filter hardware. We also discuss investigations of the performance of this system in real-time tracking of nonstationary signal sources and in extracting time-varying human vocal-tract resonances.
Optimal nonlinear transformation of LPCs to improve quantization properties
Authors: Itakura F.
Published: October 2005

It has been observed experimentally that quantization properties of linear predictor coefficients (LPC) depend on a particular point of LPC space contained in the stability boundary. Generally, the effects of quantization increase as a point in LPC space approaches the stability boundary. In this investigation, quantization properties of the usual LPC, {ai}, and PARCOR coefficients, {ki}, are analytically studied in terms of a spectral sensitivity measure, which is derived from the likelihood function of an all-pole stochastic signal. It is shown that optimal nonlinear transformation of {ki} (i = 1,…,p) which makes the sensitivity measure independent of the transformed values of parameters should be of the form zi = ci(k1,…,kp) arcsin (ki). This result is consistent with previous findings in that the optimal transformation should be an expanding characteristic. This transformation of the arcsin type gives uniformly favorable quantization properties, especially when the ki approach unity in magnitude.
Observation of the low-frequency (1 kHz) relaxation in seawater using a 200-liter glass spherical resonator
Authors: Simmons V. P. , Fisher F. H. , Levinson Stuart A.
Published: October 2005

In an effort to examine the 1-kHz relaxation observed in the ocean, a 200-liter glass sphere has been modified to operate as an acoustic resonator. By varying the temperature of the resonator, a 1-kHz relaxation can be observed. Data obtained using artificial seawater in the resonator indicate that boric acid may be a primary contributor to the low-frequency sound absorption noted in the ocean. This observation is supported by T-jump work using boric acid in a 0.725M NaCl solution. The T-jump relaxation times are definitely pH dependent over the pH range 7–9 and indicate some concentration dependence. [This report is the result of research sponsored by the Office of Naval Research.]
Interpretation of low-frequency attenuation in a surface duct
Authors: Diachok O. I. , Long G. K.
Published: October 2005

Computed values of the attenuation coefficient, which were derived from long-range measurements of acoustic transmission loss in an ice-free surface duct in the vicinity of the Arctic ice-water boundary are analyzed in the frequency range 0.1 to 1.0 Hz. The resultant attenuation in this oceanic regime, which has also been measured by D. L. Bradley et al. [J. Acoust. Soc. Am. 52, 172(A) (1972)] with similar results, is significantly greater than the attenuation measured in the deep sound channel. The observed results are discussed in terms of three previously proposed contributions to the total attenuation, viz. (1) volume absorption loss as formulated by W. H. Thorpe [J. Acoust. Soc. Am. 38, 648–654 (1965)], (2)attenuation by sea-surface scattering loss, as derived by H. W. Marsh, M. Schulkin, and S. G. Kneale [J. Acoust. Soc. Am. 33, 334–340 (1961)], and (3) a depth-dependent diffusion loss as postulated by K. M. Guthrie et al. [J. Acoust. Soc. Am. 55, 545(A) (1974)]. The surprisingly good agreement between the observed and the computed magnitudes of attenuation appears to lend some support to the hypothesis that volume attenuation is depth dependent.
Acoustic wave propagation across an oceanic front
Authors: Barakos P. A. , Speidel S. L.
Published: October 2005

The parabolic approximation to the wave equation (Tappert, 1974) is employed to compute the two-dimensional low-frequency underwater acoustic field in a surface duct interrupted by an oceanic front. The sound speed along the propagation path was calculated from almost synoptic temperature profiles obtained from a towed thermistor chain (Cummins, 1972). A comparison is made between the calculated and the observed acoustic field at two frequencies and five receiver depths. The numerical technique is then used to calculate the acoustic field for other source-receiver geometries and to establish the importance of horizontal variability in the sound-speed profiles and look into the effects of the oceanic front upon acoustic wave propagation.
Time reversal of SOFAR arrivals
Authors: Del Balzo D. R. , Moseley W. B.
Published: October 2005

Ray theory techniques are used to investigate a time-reversal phenomenon seen during the KIWI-ONE (Phase II) propagation experiment. Shots detonated along a track between New Zealand and South America were received by a deep suspended hydrophone. For ranges less than 3000 km, the standard SOFAR energy crescendo and abrupt termination is apparent, but for longer ranges, more than half of the received energy arrives after the strong axial arrival. This unexpected time reversal is shown to be cause by a broad minimum layer in the sound-speed field beyond 3000 km. Computer ray traces show that this broad minimum layer can produce the 1 sec of time reversal seen at 5300 km, while a more nearly classic SOFAR profile measured at the receiver exhibits negligible reversal. Traces simulating KIWI shot signatures were performed through a range-dependent sound-speed field with a linear segmented bottom. The calculated arrival structure agrees well with the data in both time and level. Fine structure of the arrival-time distribution is shown to correspond to discontinuities in the gradient in the broad minimum layer. [Supported by ONR 412.]
Long-range arrival-time distribution for a trilinear profile
Authors: Warfield J. Thomas
Published: October 2005

Shot signatures received in deep water near the Galapagos Islands during the KIWI-ONE (Phase II) long-range propagation experiment show much of the acoustic energy arriving after the strong axial arrival, in marked distinction to classic SOFAR reception. This has been attributed to the presence of a broad minimum layer in the measured profile [D. R. Del Balzo, J. T. Warfield, and W. B. Moseley, “Time Reversal of SOFAR Arrivals,” NRL Memorandum Rep. 2868 (1974)]. In this talk, the broad minimum layer profile is idealized by replacing an interval surrounding the channel axis of a bilinear profile with either a weak-gradient or an isospeed layer. The presence of the inserted layer causes ray arrival time to undergo a transition from one SOFAR-type distribution to another of lower channel speed as ray vertex speed cv increases. Thus the axial arrival (minimum cv) can be received earlier than some rays of higher cv. Examples illustrate the effect on the arrival-time distribution of the thickness and gradient of the inserted layer. Application to KIWI is discussed briefly. [Supported by Office of Naval Research, Code 412.]
Computer-implemented acoustic channel model
Authors: Atkinson James E.
Published: October 2005

A computer-implemented acoustic channel model has been developed to provide simulation of the effects of a wide range of ocean characteristics on underwater acoustic communications signals. The model permits evaluation of various communications systems at a much smaller cost and with a greater degree of control and repeatability than is possible through at-sea testing. It has been implemented by using FFT techniques to convolve a given input signal with a predefined channel impulse response. The model is flexible enough to accommodate a wide range of time smear and multipath effects normally found in the ocean. It permits evaluation of both existing and proposed communications systems in order to establish their performance under various operational conditions. The model has been tested with pure-tone, voice, and scrambled-voice transmissions. Specific results are presented for voice signals under conditions of time smear and multipath distortion.
The inner ear as a biochemical system considered from an otopathological point of view
Authors: Hawkins J. E.
Published: October 2005

By long tradition students of hearing have been more concerned with the biophysical than with the biochemical aspects of inner-ear function. The discovery in 1954 by Smith, Lowey, and Wu of the striking differences in the concentrations of major cations between perilymph and endolymph indicated that specialized mechanisms of membrane permeability and active transport must be involved in the regulation of their respective compositions. Also implicit in that finding was the idea of the blood-labyrinth barrier and the microhomeostasis of the inner ear. The development of sensitive microanalytical methods and the emergence of ototoxicity as a hazard of antibiotic and diuretic therapy have given additional impetus to biochemical studies of cochlear tissues and have underscored similarities of structure, function, and drug susceptibility between the stria vascularis and the epithelium of the kidney tubules. Changes in the stria occurring after noise exposure and before hair-cell degeneration support the hypothesis that the immediate cause of noise-induced threshold shifts is a derangement of the metabolic and regulatory activities of this essential cochlear tissue. Aminoglucoside accumulation in inner-ear fluids (Stupp) and delayed dye clearance after noise exposure (Schnieder) suggest a mechanism of interaction and mutual potentiation between otopathogical processes.
Experimental alteration of inner-ear fluid homeostasis
Authors: Juhn S. K.
Published: October 2005

The maintenance of a constant chemical composition of inner-ear fluids seems to be necessary for the consistent conversion of vibratory energy into a neural impulse in the labyrinth. The rate of change in concentration of various solutes in perilymph was determined in various animals after experimentally induced acidosis, hyper- and hypoglycemia, hyper- and hypocalcemia, and after adrenal cortical insufficiency. In addition, the chemical composition of perilymph, cerebrospinal fluid (CSF), and blood was studied after intracisternal administration of urea, albumin, and ouabain. It was noted that perilymph solute concentration was altered by changes in blood or CSF composition. The data obtained suggest that mechanisms may exist which control the exchange of substances between the perilymphatic space and the other fluids studied. The rate of exchange was found to be dependent upon various properties of the solute, including molecular weight, polarity, and lipid solubility. Mechanisms which participate in maintaining homeostasis of inner-ear fluids would appear to be necessary for normal labyrinthine function. The available data support the concept of a blood-labyrinthine barrier which participates in the maintenance of inner-ear fluid homeostasis.
Kinetic description of sound absorption in the atmosphere
Authors: Bass H. E. , Piercy J. E. , Evans L. B. , Sutherland L. C.
Published: October 2005

Utilizing kinetic information obtained from acoustical, optical, and opto-acoustical experiments, a comprehensive description of molecular relaxation in the atmosphere has been developed as the theoretical basis for a method of calculating the absorption of sound in air discussed in the following paper. The effects of atmospheric pressure and sound frequency on absorption evolve easily from the theory. The limitations imposed on theoretical calculations by insufficient energy transfer rate information are discussed. Many of the rates required to predict sound absorption are most easily measured using optical techniques. The similarity of acoustic and high-energy laser beam propagation through the atmosphere is discussed. [Work supported by the Army Research Office, Durham, North Carolina.]
Responses of a digital-computer model of the cochlear partition
Authors: Hubbard A. E. , Geisler C. D.
Published: October 2005

A digital-computer representation of the cochlear partition has been developed based on the model of Lein [Monograph 212, Biomedical Computer Laboratory, Washington University, St. Louis 1973]. First, a discrete-element representation of the model was developed, which is of the Zwislocki type with each resistance replaced by an infinite R-L ladder network. Suitable approximations were obtained with severely truncated networks. Secondly, a 100-section digital-computer representation was developed, using an adaptive, time-sampled, iterative algorithm that runs at approximately 10 samples per cycle for the highest frequency involved. Steady-state results will be compared with those computed by Lein using the Hamming method. The model was further modified to have lower, nonlinear resistances. The output of the modified model will be compared to experimental data obtained by Rhode and Robles [J. Acoust. Soc. Am. 55, 588–596 (1974)].
Low-level response phases of auditory nerve fibers
Authors: Sokolich W. G. , Zwislocki J. J.
Published: October 2005

The activity of single auditory nerve fibers was recorded at low response levels. Phase data obtained from both kanamycin-treated and untreated Mongolian gerbils were corrected by the average middle-ear transfer function, and the phase of basilar membrane displacement corresponding to maximum fiber excitation was determined. Units encountered in the treated animals and having abnormal phase functions showed a constant phase of maximum excitation over the frequency range where wave propagation is believed to be nondispersive. Units with CFs corresponding to cochlear regions where only inner hair cells were present showed maximum excitation near the peak displacement toward scala vestibuli. In regions where outer hair cells are believed to predominantly control the fiber responses, peak excitation occurred during displacement toward scala tympani. These results are consistent with our earlier findings of phase opposition between the inner and outer hair cells and with our model that accounts for single-unit tuning curves in terms of cancellation between the inputs from inner and associated outer hair cells. The agreement between predicted and measured phase functions of units encountered in untreated animals adds further support to the model's validity.
Computer simulation of auditory nerve characteristics: an interactive program for undergraduate laboratory instruction
Authors: Carrier Sam C.
Published: October 2005

A computer program, written in BASIC and implemented in a time-sharing environment, allows students to explore the response properties of simulated VIIIth-nerve neurons. The program has proven useful in undergraduate laboratory instruction. It allows rapid data collection and requires students to confront problems of analyzing and interpreting electrophysiological data. Thus, students can develop analytic skills without the technical competence, time-consuming effort, and expense of actual electrophysiological experimentation. Students must select the parameters (frequency and amplitude) and determine the number of repetitions for each stimulus. The program then prints the number of neural impulses elicited by each tone burst. The 12 prototype neurons are characterized by (1) best frequency, (2) threshold, (3) shape of tuning curve, (4) firing rate-vs-stimulus amplitude relations, and (5) mean and variance of spontaneous activity. After several hours of data collection most students are able to plot tuning curves and discuss how each neuron responds to different amplitude stimuli; they are somewhat less successful in describing how these properties could account for psychophysical data. Student evaluations have shown this to be the most popular laboratory of the course, since the interaction task requires rapid development of hypotheses and provides immediate feedback.
Two-tone suppression and combination-tone driving
Authors: Greenwood D. D. , Merzenich M. M. , Roth G. L.
Published: October 2005

Unit responses in the AVCN of the cat have been studied using two- and three-tone stimulus combinations. With two-tone stimulation, f1 was placed at the unit's best frequency and a higher tone (f2) added at an equal or greater intensity. In these preliminary experiments, maximum suppression of the f1 response was observed when the frequency of f2 was located at a position 0.5–0.8 mm from f1 on the basilar membrane. This distance did not appear to vary with the best frequency of neurons and might be related to critical bandwidth. During three-tone stimulus experiments, f1 was set at the unit's best frequency and f2 and f3 added above the unit's tuning curve. The frequency separations between the three tones were equal; hence f1 and 2f2 − f3 both fell at the unit's best frequency. The phase relations of the three tones could be systematically varied by varying the phase of f1. The neural responses generated by the resultant of f1 and 2f2 −f3 allowed for the determination of their relative amplitudes. These data have revealed that the amplitude of 2f2 +f3 is actually less than would be expected from cancellation-tone estimates. The reason is that in cancellation experiments the primaries suppress the cancellation tone (f1 in the present experiments). Data indicate that the amplitude of 2f2 −f3 is maximal when the intensities of f2 and f3 are equal, and that the amplitude decreases markedly as the intensity of f3 is increased. Finally, the three frequencies were shifted to place the resultant combination tone, 2f1 −f2 and 3f2 −2f3, at the best frequency. Variation in the phase of f1 produced alteration of the neural response to this resultant. These data appear to parallel the psychophysical results of Goldstein.
Properties of responses to tones and noise of single cells in the dorsal cochlear nucleus of decerebrate cats
Authors: Young Eric , Brownell William E.
Published: October 2005

Inhibition is a much more prominent feature of the discharge patterns of single cells in the dorsal cochlear nucleus (DCN) of unanesthetized cats than in the DCN of cats anesthetized with Nembutal [Evans and Nelson, Exp. Brain Res. 17, 402 (1973)]. We studied the response properties of single cells in the DCN of decerebrate cats. The units studied were histologically localized relative to the three layers of the DCN. Roughly half the DCN cells encountered showed tuning curves to short (200 msec) tones consisting of a central V-shaped inhibitory region flanked at lower and higher frequencies by thin bands of relatively weak excitation. The excitatory bands meet below the central inhibitory region at the best frequency of the cell; their prominence varies from unit to unit. The response in the central region is mixed, usually consisting of an onset discharge, followed by suppression of discharge; large after-discharges are frequently seen. In response to longer tones (3–10 sec), some traits continue to show inhibition; others begin to fire over a time course of several seconds. These units are most common in the fusiform/granule cell layer, but are also encountered elsewhere in the DCN. Despite their predominantly inhibitory response to tones, these cells may respond vigorously to broad-band noise. Unlike the situation usually described for DCN neurons, these units respond more vigorously as a band of noise centered at best frequency is progressively widened. [Supported by NIH.]
Interdisciplinary approach to combating noise pollution
Authors: Dickman Donna McCord , Wallenberg Ellis A.
Published: October 2005

Effective control of the effects of noise on people and the environment requires the cooperative effort of professionals in several disciplines. The purpose of this paper is to describe and assess an ongoing program of this type. In May 1973, funded by a grant from the United Way of Metropolitan Washington, D. C., the Washington Hearing and Speech Society, a nonprofit organization, initiated a hearing conservation program which combined the professions of audiology, engineering, medicine, and law in a cooperative effort to aid the metropolitan area's community, government, and industry in the assessment and management of environmental and occupational noise problems. In its first year, the program has contributed to (1) the initiation of numerous industrial hearing conservation programs, (2) the development of prototype models for government agencies, (3) increased community awareness of the health effects of noise and how to combat it, (4) training audiometric technicians for government and industry, and (5) interest in local noise-control legislation. It seems evident that the ever-increasing concern with the problems created by noise pollution will result in a growing need for additional multi-disciplinary programs. Thus, the examination of the development and activities of an ongoing program may well be of particular interest to members of the Acoustical Society who wish to initiate similar programs.
Environmental noise monitor
Authors: Schomer P. D. , Konheim A.
Published: October 2005

An environmental noise monitor has been designed, constructed, and tested for the U. S. Environmental Protection Agency. The system includes a Brüel & Kjaer 4921 outdoor microphone system, a Wang 600-14 computing calculator as the digital controller and statistical analyzer, and a Nagra SJ tape recorder for analog recording. The system has a 80-dB dynamic range and is capable of separately classifying data acquired during the period of time in which the winds are above a preset value. The digital system constructs variable length blocks of data. For each block various L levels as well as Leq are calculated. Running totals are kept of the daytime and nighttime Leq and each 24 hours a new Ldn is calculated. By using 3-hour blocks, the system can run unattended for a week at a time. Because the calculator is part of the system, results are always immediately available in the field.
Studies of noise propagation and atmospheric attenuation using instrumented towers
Authors: Hilton D. A. , Kasper Pete
Published: October 2005

The purpose of this paper is to describe the main features of a comprehensive research program whose overall objective is to establish the effects of the random characteristics of the atmosphere on the propagation of aircraft noise into the airport community. Included in the paper will be a description of the initial project under this program and a discussion of preliminary results. A joint NASA/NOAA/Wyle Laboratories project utilizing a noise source fixed atop the NOAA 500-ft weather research tower in Haswell, Colorado, has just been completed and represents the initial step in the long-range program. Acoustic measurements were made by locating microphones at different distances from the source along one of the tower guy wires. Existing tower-mounted weather instrumentation was used to obtain a description of the atmosphere at several locations above the ground. In addition, indications of other atmospheric parameters such as Cv and CT were obtained through the use of the NOAA bistatic echo-sounding array and the NCAR boundary-layer profiler. Preliminary data indicate that, for certain atmospheric conditions, differences in both the mean and fluctuating noise level of up to 10 dB were observed between microphones located 200 ft apart.
Blast noise sound propagation measurements
Authors: Schomer P. D.
Published: October 2005

Seven hundred 5-lb charges were set off at ground level and simultaneously recorded in four directions at distances of 1000, 2000, and 5000 feet and 2, 5, 10, and 15 miles. A specially equipped airplane gathered meteorological data, including wind speed, wind direction, temperature, humidity, and turbulence as a function of altitude. Complete radio communications ensured that recorders were turned on and off at the proper times. The purpose of the experiment is to correlate blast noise propagation including the associated blast spectra with various meteorological conditions. Initial results are presented.
Sound attenuation over homogeneous ground cover
Authors: Bolen L. N. , Bass H. E. , Lucy E.
Published: October 2005

The attenuation of 25- to 250-Hz sound waves has been measured over a homogeneous ground cover. One-third-octave bands of pink noise were transmitted across a field of soy beans and sound-pressure levels were recorded as a function of frequency, distance of absorber, and height and density of the absorber. The total absorption was compared to theoretical predictions of attenuation of sound waves layered boundaries. [Work supported by the Army Research Office, Durham, North Carolina.]
Musical acoustics: Seashore and Iowa 1907–1947
Authors: Small Arnold M.
Published: October 2005

Carl Emil Seashore, Ph.D., Yale '95, was a dominant figure at the University of Iowa for nearly the first half of the 20th century. As Professor and Head of the Psychology Department and Dean of the Graduate School, his innovative leadership led to application of the scientific method to the study of music, art, and speech among other fields. The emphasis in music was principally psychoacoustic in nature and resulted in research studies spanning 30 years, with more than one hundred major publications by the scientists involved. The scientific rationale and aesthetic constructs concerning the studies in musical acoustics are presented and critically evaluated. Contributions to musical acoustics at Iowa during that period are synoptically reported, discussed in the light of later developments, and compared to current musical practices. The presentation encompasses (1) measurement of musical talent, (2) studies of vocal and instrumental vibrato, (3) analysis of musical performance, (4) research on tone quality, and (5) research on tone production.
An illusion with musical scales
Authors: Deutsch Diana
Published: October 2005

A dichotic tonal sequence was constructed, consisting of the repetitious presentation of the C major scale, with successive tones alternating from ear to ear. The scale was presented simultaneously in both ascending and descending form, such that when a component of the ascending scale was in one ear, a component of the descending scale was in the other, and vice versa. Various illusory percepts were obtained, which differed statistically for the two-handedness populations. About 50% of the righthanders obtained a localization illusion, whereby the upper tones of the dichotic sequence were all perceived as emanating from the right earphone, and the lower tones from the left. When the earphone positions were reversed, this percept was maintained with the paradoxical consequence that the earphone which had apparently been emitting the high tones now appeared to be emitting the low tones, and vice versa. Further analysis revealed that this illusion is based on a mechanism different from that underlying the dichotic illusion recently reported by Deutsch [J. Acoust. Soc. Am. 55, S18 (1974)].