bluuebunny/crossref_metadata_2025 huggingface dataset:
Experiments concerning Near-Field-Noise Filtering
Authors: Lord J. B.
Published: September 2005

Parvulescu [J. Acoust. Soc. Am. 33, 1681(A) (1961)] has shown theoretically that proper combinations of hydrophone arrays and electronic components can be employed to receive a distant acoustic source while rejecting a near-field source. After briefly reviewing the theory, this paper describes some underwater experiments confirming Parvulescu's concepts, and discusses applications and limitations of the techniques. The experiments were conducted using both pulsed carrier and continuous broad-band signals.
Matched Filter Detectors for Additive Arrays or Receivers near Reflectors
Authors: Clay C. S. , Frosch R. A. , Liang W. L.
Published: September 2005

A matched filter operates on a signal to yield an output that is the autocorrelation of the input signal. To study the use of matched filters with an additive array, we did the following experiments: A loudspeaker was driven with an impulse, and the sound observed with an array of microphones. The electrical outputs of the microphones were added without time delays to yield a single channel of information. The matched filter-shift register delay line was adjusted to match the signal from the array when the array was in the end-fire direction. The array-matched filter combination had a correlation maximum for the direction in which the array output and filter were matched and the maximum decreased as the array was turned away from this direction. A single receiver near several reflectors also yields several arrivals on a single channel of information. The balance of the experiment was similar to the array-matched filter experiment. [Columbia University, Hudson Laboratories Informal Documentation No. 15. This work was supported by the Office of Naval Research under Contract Nonr-266(84).]
FLIP Operational Trials
Authors: Fisher Frederick H.
Published: September 2005

The 600-ton, 355-ft-long, manned ocean buoy FLIP has successfully passed its initial trials at Dabob Bay, Washington, in which it has been flipped from the horizontal to the vertical position, in which it has a 300-ft draft, and back again to the horizontal position. Its natural heaving period was 27 sec, as expected. The flipping operation is discussed, and films are shown. Records of the motion during the flipping operation made by the 3D range facility of the Keyport Naval Torpedo Testing Station are shown. In going from the horizontal to the vertical, the time to change from 10° to 80° is about 100 sec. [This presentation represents results of research sponsored by the Office of Naval Research.]
Hydroelasticity and Noise Control
Authors: Bland Richard E.
Published: September 2005

Hydro-elasticity is the study of the interaction relationships between inertia forces, hydrodynamic forces, and elastic forces acting on structures as they are propelled through the water. The nomenclature and theory stem from the discipline known as aeroelasticity, the major differences being due to the density ratio between air and water, i.e., the medium in which the structure is submerged. The Garfield Thomas water tunnel at the Ordnance Research Laboratory offers a unique facility for studies in this discipline. These studies include flutter, flow noise, radiation, and the generation of exciting forces by flow over appendages and through propulsors. The experimental approach utilizes simple structural modeling, some of which are shown and discussed. A limited amount of data on body response and hydrodynamic forcing functions is presented to demonstrate present techniques and to establish validity of the model approach.
Measurements of Correlation Properties of Fully Developed Turbulent Pipe Flow
Authors: Carey George F. , VonWinkle William A.
Published: September 2005

Certain properties of the turbulent pressure field at the wall of a cylindrical pipe have been investigated experimentally over a range of Reynolds number from 100 000 to 300 000. In particular, data have been obtained concerning the convection velocity and the longitudinal space-time correlation of the field both over broad frequency bands and in octave bands. Longitudinal and lateral space correlations have been measured over broad frequency bands as functions of the spatial separation and in octave bands as functions of the Strouhal number. The convection velocity is found to decrease as the frequency increases. The longitudinal space-time correlation measurements indicate that the high-frequency components decay at a substantially faster rate than those of the low-frequency components. The data obtained for various spatial separations, frequency bands, and flow velocity appear to define a single lateral space-correlation curve and a single longitudinal space-correlation curve. In addition, estimates of the longitudinal and lateral scale as a function of frequency band have been obtained.
Calculation of Articulation Index from White-Noise Data
Authors: Stuntz Stephen E.
Published: September 2005

Short-duration samples of white noise were transmitted through 3 voice-communication systems and reduced to visual spectrum displays (sonagraph sections). Likewise, samples of residual noise were taken from each system during no-signal intervals and similarly displayed. By means of a transparent frequency-amplitude overlay scale, S+N/N ratios were determined at the midfrequencies of the octaves between 150 and 4800 cps. These ratios were used as the basis for calculating an articulation index (AI) for each system, after the octave-band method of Kryter. PB-word-articulation scores corresponding to the calculated AI's were taken from published curves; these scores compared favorably with scores obtained from tests made by transmitting PB wordlists through the three systems. Suggestions are made for refining and simplifying the octave noise-band technique.
Tactual Transmission of Speech Sounds to the Deaf. Comparison with Lip Reading
Authors: Pickett J. M.
Published: September 2005

This paper describes tests of a special analyzer-vocoder, which presents the output as a spatial array of 10 tactual stimuli to the fingers of the receiving subject. Each stimulus consisted of a 300-cycle vibration, controlled in amplitude according to the energy in the associated frequency channel of the vocoder. The channels were contiguous and centered at 210, 400, 580, 830, 1050, 1800, 2250, 3320, 5800, and 7700 cps. The vocoder was compared to lip reading as a means of transmitting selected speech sounds to profoundly deaf children. It was found to be superior to lip reading for distinguishing certain speech sounds that are difficult to lip-read, such as /m/ and /b/; the device also provided sufficient information to allow partial identification of sounds that are easier to lip-read, i.e., /s/ and /t/. Combined reception, by both lip reading and tactile stimulation, was found to be superior to lip reading alone. [Research performed during a fellowship from the National Institute of Mental Health, National Institutes of Health, Public Health Service, U. S. Department of Health, Education, and Welfare.]
Derivation and Application of the Fokker-Planck Equation
Authors: Caughey T. K.
Published: September 2005

The Fokker-Planck equation is derived and applied to discrete nonlinear dynamic systems subjected to white-noise excitation. For the class of problems in which the nonlinearities involve only the displacements of the system, it is shown that exact solutions can be constructed for the stationary Fokker-Planck equation. It is further shown that, if steady-state solutions exist. they are unique.
Empirical Evidence for Nonlinearity and Directions for Future Work
Authors: Lyon Richard H.
Published: September 2005

The topic is introduced by reviewing a selected set of examples of field data for which nonlinearity is presumed to be responsible for some sort of anomalous response behavior. This behavior may take the form of a nonproportional response-excitation effect, instability of response, spectral distortion, or peculiar probability density. Some results of laboratory experiments are also recalled; some of these were intended to model the structure for which field data were measured; some were designed to approximate a theortical model in order to test theoretical conclusions. The necessity for carefully distinguishing between laboratory and field behavior is emphasized. Successful attempts at theoretical study of nonlinear, random vibration problems are discussed, emphasizing the probable causes of success. The probable course of future theoretical study is outlined, as well as a listing of problems for which solutions are needed. A re-examination of the role of the experiment is made. It is suggested that experimental analysis should and can occupy a more central role in the study of nonlinear structural response, but that the manner in which the laboratory experiment is related to both the field measurement and the theoretical study may need to be re-examined.
Results of Some Measurements on the Reflection of Sound from Wood-Slat Surfaces
Authors: Richards R. L. , Towne R. M.
Published: September 2005

This paper presents some preliminary data on the “back” reflection of an acoustic pulse from a variety of wood-slat surfaces. The relative amplitude of the wave reflected back towards the source is shown as a function of slat configuration, angle of incidence, and backing material. Anticipated improvements in instrumentation and future analysis programs are discussed. The measurements were made in an anechoic room at the University of Washington.
Audio-Message Assembly on the IBM 7090 Computer
Authors: Rew Robert P.
Published: September 2005

This paper discusses a program, written for the IBM 7090 computer, which will assemble and deliver audio messages. The purpose of the investigation was to provide a voice output for applications where it is required, such as telephone inquiries. The main part of the paper deals with a method for taking spoken words of arbitrary length, and compressing or expanding them to fit a given time interval without affecting their pitch or quality. For example, a word of 450-msec duration could be compressed to 300 msec or expanded to 600 msec and still retain its original quality. (A recorded demonstration of this ability is presented.) Also discussed are the problems encountered in getting analog information into and out of a digital machine.
“Harmonic Compression”—A New Method for Speech Bandwidth Compression
Authors: Schroeder M. R. , Logan B. F. , Prestigiacomo A. J.
Published: September 2005

A “harmonic compressor” is described which divides the frequencies of individual harmonic-frequency components by 2, transmits the compressed signal, and resynthesizes a close replica of the input speech signal by frequency doubling. The new device has also been used to speed up or slow down the syllabic rate of speech. Application of harmonic compression to the baseband of a voice-excited vocoder has also been studied. Tape recordings of the outputs of a harmonic compressor and a harmonic compressor combined with a voice-excited vocoder are demonstrated. Samples of speeded-up and slowed-down speech are also demonstrated.
Single-Channel Speech Interpolator for 2:1 Bandwidth Reduction
Authors: Flanagan James L. , Schroeder Manfred R. , Bird Carol M.
Published: September 2005

A method is described for extending the effective frequency range of a half-band speech-transmission channel. The method capitalizes upon the silent intervals occurring in a single speech signal. Such intervals arise from stop-consonant gaps and pauses between syllables, words, and sentences. The single speech signal is filtered into equal high- and low-frequency bands. The low band is transmitted directly and stored at the receiver. The high band is delayed at the transmitter, heterodyned down to the low-frequency range, and transmitted during the available gaps. The gaps generally are shorter than the bursts, so that some of the high band will be lost. At the receiver, delays are equalized and the bands reassembled. A burst-gap timing signal is transmitted separately, but requires small bandwidth. The price of the 2:1 bandwidth reduction is a constant, but tolerable, transmission delay and some loss, usually small, of the high band. Brief, separated utterances are transmitted perfectly. Prolonged gapless passages are never degraded beyond the half-band condition. The transmission system was completely simulated and tested on an IBM 7090 digital computer. A taped demonstration of speech transmitted by the system is presented.
Determination of Loss Factors of Materials in Powder and Granular Form
Authors: Simon Murray M.
Published: September 2005

This study concerns the determination of loss factors of materials in powder and granular forms by means of decay-time and bandwidth measurements. Two types of liquid-filled tubes are used for the experiment. One is a heavy-walled stainless-steel cylinder driven by a steel membrane. The other is a glass U tube driven by a loudspeaker above the liquid column. The loss factors are determined for the liquid-filled tubes. The material to be studied is then placed at a velocity node within the liquid, and the loss factor of the combination is determined. From these determinations, the loss factor of the additional material, as well as its bulk modulus of elasticity, can be calculated. [This research is supported by the U. S. Air Force Office of Research.]
Mutual Interaction between Two Spheres in a Plane Sound Field
Authors: Embleton T. F. W.
Published: September 2005

Mutual forces between 2 spheres having their lines of centers parallel to the direction of propagation of traveling plane waves may be many times greater than the usual forces of repulsion exerted on a single sphere. Numerical calculations show that the mean force on the sphere further from the source of the sound field is always one of repulsion, but increases in magnitude as the spheres are brought closer together. The mean force acting on the other sphere oscillates with increasing amplitude as a function of decreasing separation between the spheres, reaching a maximum value of repulsion from the source when the separation between centers is a quarter wavelength. At smaller separations, mutual repulsion between the spheres is the dominating force. This same behavior is predicted over a large range of sphere radii a and wavelength λ—at least from 2πa/λ=0.01 to 1.0.
Closing the Binaural Gap
Authors: David Edward E.
Published: September 2005

Results from binaural earphone and free-field experiments have remained somewhat distinct. The former typically utilize well-controlled elementary stimuli, and have yielded numerous and precise data. These have been a major basis for inferring physiological models of auditory mechanisms. On the other hand, many of the most consequential binaural effects are free-field phenomena, for instance, auditory perspective, stereophony, the precedence effect, and the cocktail party effect. Here, stimuli are more difficult to control and are typically more complex. As a result, free-field experiments have often led to data not easily reproducible nor easily related to basic auditory mechanisms. A few pioneering efforts to span this gap were made by Békésy, Wallach, Rosensweig, Newman, and perhaps others. Recent work emphasizes that free field binaural hearing involves factors beyond those ordinarily explored in earphone experiments.
Measurement of the Sound Velocity and Loss Factor of Polystyrene from 60 kc to 1 Mc by a Pulse Echo Technique
Authors: Lange James N.
Published: September 2005

In the frequency range from 60 kc to 1 Mc, the boundary conditions at the free surfaces are very complex. A diliational wave reflected from a boundary such as the ends of a rod may be decomposed into a shear wave, which is generated upon reflection of the diliatational wave, and a diliatational wave. The amplitude of the reflected diliational wave is decreased due to the energy conversion upon reflection as well as the attenuation of the material. The measured loss factor is, therefore, not indicative of the energy absorption of the material. In order to obtain an accurate measurement of the absorption, specimens in the shape of cones, hemispheres, and spheres were investigated. Corrections were also made for effects of dispersion. Identification of the modulus associated with the sound velocity in a frequency range above and below the dispersion region was attempted. (This work was supported in part by the U. S. Atomic Energy Commission.)
Hydrophone Calibration in a Vibrating Column of Liquid
Authors: Schloss F. , Strasberg M.
Published: September 2005

Absolute pressure calibrations of hydrophones have been performed in the laboratory by immersing the hydrophone in a column of liquid set into vibration. Sound pressures of over 1000 dyne/cm2 can easily be obtained. Various limitations on the procedure, such as those due to fluid motion around the hydrophone, and wave effects in the liquid column are discussed.
Measurement of Seiche Activity in San Diego Bay
Authors: Collins Joseph L. , Ellis Glen E.
Published: September 2005

Recordings of bottom-pressure fluctuations were made in San Diego Bay in April 1960 and in January 1961 to study the seiche phenomenon. A solion infrasonic hydrophone was utilized as a bottom-pressure transducer to detect pressure fluctuations over the range of 5- to 1600 sec periods. Analysis of the bottom-pressure records consists of computing relative power spectra estimates and autocorrelation functions after the raw pressure data are handpass filtered. From the relative power spectra and limited theoretical considerations, it was determined that there is an active longitudinal seiche with a second harmonic period of 1450 sec. A description of the solion infrasonic hydrophone is presented along with the recording and analysis system.
Computation of Far-Field Characteristics of a Transducer from Measurements Made in a Reflective Tank
Authors: Baker D. D.
Published: September 2005

The directivity pattern and source level of a large cylindrical transducer were computed using the Kirchhoff formula with a simple approximation for the normal pressure gradient. The integration was performed numerically over a finite cylinder. Nearfield pressure amplitude and phase data were made over the surface of integration by rotating the transducer and measuring the output of a line hydrophone less than one wavelength from the active face. These measurements were made in a highly reflective sheet-steel tank of diameter 2.9 times that of the transducer and wall thickness much less than a wavelength. The amplitude and phase were measured on the leading portion of the received pulses. Directivity patterns were computed in a plane normal to the axis of the transducer. They were compared with the equivalent patterns measured in open-sea conditions. Agreement was good both in pattern shape and source level. It is felt that this work shows that it is possible to calibrate such a transducer in a tank that is relatively small and highly reflective.
Achieving Transducer Pressure Release with Compliant-Metal Tubes
Authors: Miller Norman D. , Vincent Gordon A.
Published: September 2005

High-tensile steel tubing, when preformed to an elliptical cross section, can provide pressure release for high-power, low-frequency, underwater-sound projectors up to depths where the hydrostatic pressure stresses the tubes beyond the yield-strength limit. The compressibility of the compliant tube is essentially constant up to the point where the material yields. When using compliant tubes for pressure release, the volume of tubes is chosen such that their volume displacement (compressibility) is equal to the volume of fluid displaced by the active transducer elements. The dimensions of the tubes are such that they are operated well below resonance; hence, a 1:1 compliance ratio exists between the driver element and pressure-release system. When the dimensions of tubes and of tube cavities are such that operation is near the resonance condition, the apparent compliance of the system increases, and a lesser number of tubes can be used to provide pressure release. Pressure-release systems, using compliant tubes made from heat-treated 4130 steel, have been built for use at depths to 2000 ft. Comparison tests between a transducer using a pressure-regulated air-release system and the compliant tube-release system have been made, and the performance of the transducer is essentially unchanged.
Optimum Lengths of Valve Tubes for Brass Wind Instruments
Authors: Young Robert W.
Published: September 2005

A difficulty in the design of brass wind instruments arises from use of the valves both singly and in combination: a single valve tube that will lower the frequency the proper fractional amount lowers the frequency a lesser fractional amount when it is added to the main tubing already lengthened by another valve tube. From a trial-and-error study, John Redfield [J. Acoust. Soc. Am. 3, 292 296 (1931)] proposed that the tones produced separately by the three valves be tuned, respectively, 16, 12, and 21 cents flat. The present paper shows how the discrepancies can be minimized efficiently by the method of least squares and in a way best suited to a given instrument. For many cornets and trumpets, a good compromise can be attained by making the tones produced by the three valves respectively 5, 1, and 16 cents flat; then, the tones that must be obtained by the three valves in combination will be about 15 cents sharp. This result is based on a statistical analysis of some 17 000 measurements of intonation of 27 instruments that indicates that the frequency of sound from a cornet or trumpet varies inversely as the 5/3 power of the length thereof. For baritone and brass horns, however, the frequency varies inversely as the length, and a different tuning rule is appropriate.
Timbre Cues and the Identification of Musical Instruments
Authors: Saldanha E. L. , Corso John F.
Published: September 2005

This study was designed to evaluate the relative importance of transients, harmonic structure, and vibrato as timbre cues in the absolute judgment of musical tones. Tape recordings were made of tones played, by experienced musicians in an anechoic chamber, on ten different instruments at three selected frequencies of the equally tempered scale. Appropriate retaping and splicing procedures provided five types of stimulus material: (1) entire tone (initial transients, steady state, and final transient); (2) entire tone (initial transients, shortened steady state, and final transients); (3) initial transients and steady-state portion only; (4) steady-state portion only; and (5) steady state and final transients only. The final test tape consisted of 300 randomly ordered tonal stimuli (10 instruments×3 frequencies×2 playing styles×5 types of tones). Twenty trained musicians were tested and retested on the final tape in two experimental sessions. Ss were provided with a reminder list of 39 instruments (grouped into classes), and were required, on hearing a test tone, to identify the appropriate instrument. The results of this study indicate that (1) some instruments are better identified than others, (2) absolute judgments are better at F4 than at C4 or A1, (3) a vibrato tone is better-identified than a nonvibrato tone, (4) initial transients with a steady-state portion of the tone produce the best identification, and (5) identification is significantly improved with practice. In general, discrimination was relatively poor, with only about 40% of the absolute judgments being correct.
Degarble
Authors: Frick F. C.
Published: September 2005

Several years ago the Lincoln Laboratory developed a special-purpose computer [MAUDE] to decode hand-sent Morse signals. Since the computer was to handle all kinds of Morse, including cipher, it took no advantage of linguistic redundancies, and the question arose as to whether the intelligibility of its output for English text could be improved by a simple computer program that would take into account some of the constraints of the language. Degarbler is an IBM-7090 program, developed by (Mrs.) C. K. McElwain, that at least partly corrects English text as it might be corrupted by a noisy Morse code channel and MAUDE decoding. The linguistic information utilized by the program is restricted to a knowledge of vocabulary and a table of the Morse-code equivalents of the alphabet and the numerals. As the program now stands, it corrects between 64% and 74% of the garbled words of messages having a 3% to 9% letter-error rate. Qualitatively, the results appear better than the percentages indicate. The program accepts very nearly unintelligible text and renders it easy to read.
Syntactic Structure of Printed Sentences
Authors: Oettinger Anthony G.
Published: September 2005

It is currently fashionable in linguistics to define language as a system of vocal symbols. This practice, and the underlying tendency to consider speech as a primordial entity of which writing and other notations are at best poor mirrors, is natural enough when studying languages that have no writing systems or when the nature and transmission of speech sounds are a primary concern. Writing, however, is the primary medium for the bulk of scholarly and scientific communication. Some languages or subsets of languages are hardly ever spoken—witness the notations of mathematics or of organic chemistry. The study of a nonvocal symbol system is, therefore, of some importance. Most linguists would agree with Hockett that, “Speech and writing are not merely two different manifestations of something fundamentally the same.” But, surely this is a matter of degree. If and when both the speech and writing systems of a language are fully understood then, should they turn out to have essentially isomorphic structures, it would quite likely make sense to regard both as slightly varying representations of one common underlying abstract system. If, on the other hand, they should turn out to have strikingly different structures, there would be considerable justification for treating them as weakly related or altogether distinct systems. In any case, it is premature to rule out the possibility that current studies of the syntax of printed language may contribute to the solution of speech-recognition problems. A method due to Kuno for producing multiple parsings of syntactically ambiguous, printed English sentences is described. This method is based on an extension of the technique of predictive syntactic analysis as originated by Rhodes at the National Bureau of Standards and elaborated at the Harvard Computation Laboratory. It supplies alternative acceptable structures for sentences whose parsing is rendered ambiguous either by homography (membership of a given-word form in more than one syntactic word class) or by multiple, possible syntactic, functions of a given-word class. Such techniques, when proven, could materially assist in the process of speech recognition when applied to suitably prepared discrete representations of speech.
Response of Windows to Random Noise
Authors: Freynik Henry S.
Published: September 2005

A possibility of window breakage due to noise of large boosters has led to an investigation of stresses in a residential-type window exposed to random-frequency high-intensity noise. The natural frequency, structural damping, and static failure pressures were determined, as well as failure due to sinusoidal acoustic excitation. Stresses were measured at various points on the window for both random and sinusoidal excitation. It was found that the highest stresses occurred near the center of the window and at its fundamental frequency. The 3-ft-sq test window of double-strength glass showed surprising resistance to acoustic fatigue by surviving a random noise field of over 150 db for a period of 12 hr.
Atmospheric Effects on Sonic-Boom Pressure Signatures
Authors: Maglieri Domenic J. , Parrott Tony L.
Published: September 2005

Certain initial results are presented from a recent joint USAF-NASA-FAA sonic-boom flight-test program which was conducted over a specially instrumented ground range near Edwards Air Force Base, California. Sonic-boom ground-pressure signatures were obtained for a wide variety of temperature and wind profiles in the lower atmosphere. The results indicate that the signature shapes were rather insensitive to variations of the wind profile, but some marked effects were linked to thermal-induced turbulence in the lower atmosphere.
Observations on Sonic Booms and People
Authors: Guild Elizabeth
Published: September 2005

Despite much evidence from Air Shows and aircraft gunnery and bombing ranges that the shock waves emanating from aircraft flying low and at supersonic speed do not inflict direct physiological injury on persons very near the flight path, the notion persists in many quarters that sonic booms can and will hurt people. As additional metropolitan areas are exposed to booms from military aircraft at high altitude and as plans for civilian supersonic transports proceed, new rumors arise. The purpose of this paper is to report on observations, made over a period of years, of the responses of people to sonic booms. Included is a special study in which multiple exposures to the most-intense shock waves possible from current high-performance aircraft were experienced. Although startle responses are regularly evoked—their degree depending upon whether the boom is expected as well as on the individual's familiarity with the phenomenon—no direct physiological injuries are incurred from booms of the highest magnitude expected in the foreseeable future.
Noise Control for Reciprocating Compressors
Authors: Hoover R. M. , Williams L. J.
Published: September 2005

Reciprocating compressors, as used in many air-conditioning systems, have long been recognized as potential major sources of airborne and structureborne noise. However, in many installations, some of the noise paths are overlooked either because of incomplete information, ill-defined criteria, or haphazard consideration of the problems. In this paper, the need for effective noise-control measures for both the airborne and structureborne path is illustrated by several case histories involving reciprocating compressors. It is shown, however, that adequate noise-control measures can usually be provided within the framework of “normal” construction techniques.
Acoustics of the Middle Ear
Authors: Zwislocki Jozef
Published: September 2005

Recently published research on the middle ear, in conjunction with some older studies, has provided sufficient information for a quantitative analysis of the middle-ear function. In order to perform such an analysis, an electrical analog has been built that matches the middle ear element by element. It reproduces the average impedance at the eardrum measured on normal ears as well as on ears with various pathologies. Its transmission characteristic as a function of frequency agrees with data obtained on ear preparations. The analog is a convenient tool for the study of the stimulus transformation in the middle ear and of the effect of various middle-ear structures. For instance, it is possible to predict the input to the cochlea for various transient stimuli, or to investigate the effect produced by stapedius contractions. It is also possible to study the correlation between hearing loss and impedance changes at the eardrum produced by an otosclerotic fixation of the stapes.
Impedance of the Ear—A Variable Quantity
Authors: Mundie Joseph R.
Published: September 2005

The impedance of the ear at any given frequency in the auditory range is by no means a fixed or stable quantity, but may vary moment by moment over a range of magnitude as great as a factor of 10. This paper reports the results and interpretations of impedance measurements performed on guinea pigs that have led to the following conclusions. Impedance changes which occur in a short period of time appear to be associated with the conductive mechanism of the middle ear. These short-term impedance changes apparently reflect a readjustment of the conductive mechanism. They are produced by any factor that may shift the “resting position” of the tympanic membrane, such as unequal pressure between middle and outer ear (due to barometric pressure changes, swallowing, temperature changes in anesthetized subjects, etc.) or action of the tympanic muscles. At low frequencies, below 2000 cps, the tympanic membrane becomes stiffer until its impedance is about equal to that of the air in the middle-ear space. Above 2000 cps, the impedance of the total ear decreases, indicating a change in the transformation ratio between tympanic membrane and cochlea; the cochlea is apparently “decoupled” from the conductive mechanism. This is confirmed by measurement of cochlear microphonics. These facts lead to the hypothesis that tensor tympani protects the cochlea and conductive system at low frequencies by increasing the impedance of the tympanic membrane, and at frequencies above 2000 cps the cochlea is protected by the action of the stapedius muscle in disconnecting it from the chain.
Selected Topics in Nonlinear Acoustics
Authors: Westervelt Peter J.
Published: September 2005

The following topics are discussed from a unified point of view: acoustic streaming; radiation pressure and other second-order forces; scattering of sound by sound and self-scattering of sound; interaction of sound with orifices.
Nonlinear Spherical and Cylindrical Sound Waves. Nonlinear Propagation of Random Noise
Authors: Oestreicher Hans L.
Published: September 2005

The first part of this paper reviews methods and approaches which have been developed for the investigation of the quantitative laws of propagation of spherical and cylindrical sound waves with large amplitudes. The results obtained are scarcer and more restricted than those for the plane-wave case. The second part of the paper is concerned with the nonlinear propagation of random acoustic noise in a continuous medium, a problem which has found little attention in the literature. A discussion of the mathematical and physical parameters that determine the problem is given, and some theoretical and experimental results for specific cases are presented.
Historical Notes on Finite-Amplitude Sound
Authors: Blackstock David T.
Published: September 2005

Finite-amplitude sound is a topic that has enjoyed a long and rich tradition. It occupied the attention of many of the masters of applied mechanics and mathematics of the 1800's and yet is still sufficiently a mystery to hold the attention of modern investigators. This survey emphasizes the 19th-century investigations. These were devoted almost entirely to plane waves in nondissipative gases. An attempt is made to relate the individual works to the larger trends in physics. The well-known contributions of Poisson, Earnshaw, Riemann, and Rayleigh are reviewed. Attention is also given to works of Euler, Blake, Challis, Airy, and Stokes. The post-Rayleigh period is reviewed briefly.
Velocity of Cavitation Bursts
Authors: Darner Charles I. , Bobber Robert J.
Published: September 2005

A 1-in.-diam PZT-4 projector was driven with continuous power to the water-cavitation level at 1 Mc/sec. The rectified and amplified output of a probe hydrophone produced audible whistles that the literature has attributed to the formation, splitting, coalescence, resonance, or collapse of the bubbles. Repeated observations of the phenomenon correlated the audible tone with the flow velocity of the bubbles, which led to the assumption that beat frequencies were being produced between the driving freqency and a Doppler-shifted frequency reflected from the receding-cavitation burst. Further investigation showed that the beats also appeared as a low-frequency undulation of the projector-driving current and voltage which, when rectified and amplified, became audible in the form of sequential musical and discordant whistles. The use of a narrow-bandpass filter to measure sound-pressure amplitudes above and below the driving frequency confirmed the hypothesis of a Doppler-shifted frequency reacting on the sound source. Calculations showed the velocity of an individual cavitation burst to be 75 cm/sec for each multiple of 1 kc that was audible. The highest measured velocity was 10.5 m/sec.
Studies Using Spectrographic Pictures from New Speech-Analysis Instrumentation
Authors: Wood David E. , Hewitt Terry L.
Published: September 2005

In studies of the acoustic properties of speech elements, an advanced spectrum-analyzing system has been developed to display large quantities of information generated and recorded in real time. The speech signals are frequency-analyzed by passage through a large bank of filters producing outputs which are sampled at rates high enough to fully preserve the information. The frequency scanning is performed by a unique and flexible process of interpolation that clearly defines and accurately locates spectral energy concentrations. Thus, features such as the formants of vowels are consistently displayed by the analyzer, eliminating the need for additional calculations to determine their frequency locations. The spectral signals are recorded on photographic strip films to provide pictures which can be enlarged as needed for convenience in use. A variety of picture formats are available, including the familiar power spectrum vs time with intensity modulation to show the amplitude patterns. Accurate amplitude detail is provided by spectral cross sections taken at rates that produce complete, clear time histories of the changing speech relations. This paper presents speech pictures and data obtained by investigations using the new instrumentation. Results are presented showing variations in the acoustic characteristics of phonemes for different syllables and speakers.
Audio Displays of Speech Segments at Three Levels of Phonetic Context
Authors: Stowe Arthur N.
Published: September 2005

Three computer programs to segment speech have been written for the Lincoln Laboratory TX-2 computer. The segments of these programs are at three levels of phonetic context. One is at a level below that of an individual speech sound, and corresponds to individual pitch periods in voiced sounds. A second is at the level of a speech sound or short sequence of speech sounds, and is correlated to a considerable extent with place and manner of articulation [J. Acoust. Soc. Am. 32, 1516 (1960)]. The third is above the level of single speech sounds, and is closely correlated with syllables. For the last two types of segments, the following technique is used so that the individual segments may be heard separately. The speech waveform is time- and amplitude-quantized and read into the computer. From the computer, it is stored on digital tape. The digitized waveform is again read out of the computer, decoded through a digital-to-analog converter, and fed into a 35-channel filter bank. The output of the filter bank is time- and amplitude-quantized and read into the computer. This spectral information is used to arrive at segment decisions. The quantized waveform is read back in from digital tape. The segment decisions are matched up against the waveform. The quantized waveform, from beginning to end of segment, is read out of the computer, decoded, and available for recording. For the segments of pitch period length, an individual pitch period may be read out over and over, giving a continuous audio display of its harmonic structure. Some of these outputs are demonstrated, and possible uses of this display facility are mentioned.
Analysis-Synthesis of Continuous Speech in Terms of Orthogonalized Exponentially Damped Sinusoids
Authors: Manley Harold J. , Klein Doretta B.
Published: September 2005

The paper reports the results of a digital computer simulation in which a sample of connected speech was analyzed and resynthesized in terms of a series of orthogonalized exponentially damped sinusoids. It was found to be possible to synthesize each pitch period from a function set having only 16 fixed frequencies with fixed damping at each frequency. The processing was done pitch-synchronously on an IBM-7090 digital computer, using 10-bit accuracy speech which had been digitized at a rate of 12000 samples per second. Procedures for generating the sample values of the function set on a digital computer, manual pitch extraction, and data-processing methods are described. Comparisons of the input and resynthesized speech samples are presented both visually and aurally by means of a tape recording. The results indicate that both the phonetic content and the quality of the speaker's voice are retained in the analysis-synthesis process. This analysis-synthesis process, simulated in this investigation, is amenable to real-time analog realization. (This work was supported by the Rome Air Development Center under contract No. AF 30(602)-2446.)
Acoustical Suite of the Northern Electric Research and Development Laboratories
Authors: Jamroz A. , Tanner R. H.
Published: September 2005

The new Research and Development Laboratories of the Northern Electric Company in Ottawa, Ontario, Canada, have been designed to accommodate both a small auditorium (seating capacity 200) and a medium size anechoic chamber. In addition to its more normal use, the auditorium is arranged so that it may be employed as a high quality listening room for subjective tests, forming, together with the anechoic chamber and the associated laboratory, a useful acoustical facility. The paper describes the design criteria and methods behind the construction of the rooms and discusses some of the problems encountered. In the design of an anechoic chamber, these problems include the resilient suspension of the floor slab, the provision of a low-reflection surface for walking, and suspension of the inner door. Somewhat novel methods of overcoming these problems are described. Results of measurements of the performance of both facilities are included.
Acoustics of the New Fisher Theater
Authors: Klepper David L.
Published: September 2005

The New Fisher Theater in Detroit is dual purpose, designed to be used both as a 2200-seat house for musical comedy and opera and, with 600 seats in the rear of the balcony closed off by means of a sound-transparent wall, as a 1600-seat legitimate theater. Speech intelligibility for the reduced-capacity house is possible without electronic amplification; but a speech-reinforcement system is always employed for the 2200-seat house. A simple, but complete, sound-effects playback system is included. This paper describes the over-all acoustical-design objectives and the extent to which they were achieved in the finished building. The highly successful results of the new theater reflect credit upon its architects, Rapp and Rapp of Chicago, for their ability to integrate acoustical recommendations into a design that required working within an existing building and with but few major structural changes permissible.
Initial Characteristics of Impulse Response for Real and Artificial Reverberation
Authors: Harvey F. K. , Cisek J. G. , Schroeder M. R.
Published: September 2005

Studies by Schroeder et al. [J. Acoust. Soc. Am. 33, 1061 (1961); J. Aud. Eng. Soc. 9, 192 (1961)] have shown that artificial reverberation can be made both colorless and flutter-free, using computer-simulation techniques. However, for economic reasons, commercially available reverberators only approximate these desirable attributes of real reverberation, and yet are satisfactory for many sounds. Several types of reverberators have been obtained and the initial impulse response investigated and compared with that of actual enclosures. It has been found that artificial reverberation produced by these devices generally exhibits an audible pulsing action (flutter) during the first part of the reverberation process, which can be correlated with oscillograms of the impulse response. The first echo usually occurs somewhere between 15 and 50 msec after the originating pulse (direct signal). The pulse density during the immediately following period (say, 500 msec) is relatively sparse, but may build up during the remainder of the reverberation process if multiple-mode transmission occurs. In the case of real reverberation, a high-pulse density is obtained much earlier depending upon the volume of the enclosure and the positions of the microphone and sound source. The initial reverberation characteristics of devices using helical springs, tape delay, or a steel plate are compared with an auditorium, a church, and computer-generated colorless reverberation.
Sound of Change in the American Schoolhouse
Authors: King Jonathan
Published: September 2005

With schools moving away from the concept of sorting students into self-contained comparatively noiseproof boxes, each with one teacher and 30 students, towards a community of scholarship involving fluidity of space and professional relations, a wholly new approach to planning school acoustics is necessary.
Effects of Asymmetrical Masking on the Binaural Lateralization of Clicks
Authors: Flanagan J. L. , David E. E. , Watson B. J.
Published: September 2005

Brief pulses, delivered binaurally to listeners wearing earphones, produce a sound image centered in the head when the pulses are identical and simultaneous. Changing the polarity of one pulse (for example, from rarefaction to condensation) changes the relative times of occurrence that produce a centered image [J. Acoust. Soc. Am. 33, 840A (1961)]. This interaural disparity appears explicable in terms of the mechanical response of the basilar membrane and simple hypotheses about mechanical-to-neural transduction in the ear. Similarly, the disparity can be manipulated by symmetrical HP or LP masking noise. By this means, the coherent neural information is elicited from a specific place on the membrane, and reflects the mechanical properties of that place [J. Acoust. Soc. Am. 33, 1670A (1961)]. When the masking is made asymmetrical (for example, HP in one ear, LP in the other), the interaural times can be made also to reflect the transit times to specific points on the membrane. This paper presents data for lateralizing both cophasic and antiphasic pulses under the following asymmetric masking: 600 HP vs unmasked; 2400 LP vs unmasked; 600 HP vs 2400 LP. The unmasked vs unmasked condition is included as a control. The results continue to support the hypotheses advanced in the earlier work, and the behavior appears to be explicable in terms of the displacement responses of the basilar membrane.
Influence of Monaural Masking on the Localization of Sound in Space
Authors: Butler Robert A. , Naunton Ralph F.
Published: September 2005

Normally, under conditions of monaural free-field listening occasioned either by disease or by occlusion of one ear, the apparent source of a sound in space is displaced toward the side of the normal or unoccluded ear. The experiments described indicate that, under some conditions of unilateral auditory masking with pure tones or noise, the apparent source of an auditory signal in space, delivered always at zero azimuth, moves toward the masked ear. The apparent source of the signal could, in these experiments, be varied over a range of 150°, consistent changes in apparent location being produced by changes in the intensity and frequency relations between the signal in space and the masking stimulus.