bluuebunny/crossref_metadata_2025 huggingface dataset:
Minimum Phase Responses for the Basilar Membrane
Authors: Flanagan James L. , Bird Carol M.
Published: September 2005

Appreciable experimental data exist for the amplitude of basilar membrane displacement as a function of sound frequency. Data on the phase of displacement vs frequency are relatively meager. At low frequencies the latter do not relate well to the phase predicted by recent mathematical models for membrane displacement [Bell System Tech. J. 39, 1163–1191 (1960)]. To examine the difference, we programmed an IBM-7090 computer to calculate: (a) minimum phase functions corresponding to the experimental amplitude vs frequency responses; (b) inverse Fourier transforms of the experimental amplitude and phase data; and (c) inverse transforms of the experimental amplitude and calculated minimum phase data. The results were: (a) At low frequencies the calculated phase differs from the experimental phase by about π/2 rad or more, with the membrane displacement leading that of the stapes. Constant delay (i.e., linear phase) does not account for the difference; (b) inverse transforms of the experimental data yield nonrealizable impulse responses, i.e., responses which are nonzero for negative times; and (c) inverse transforms of the experimental amplitude and calculated phase yield proper impulse responses. At low frequencies, therefore, the experimental amplitude and phase data do not seem compatible. Because of the great difficulties attending the experimental measurement, it is the phase data which are thought to need confirmation.
A Mathematical Definition of Acoustic Coherence
Authors: Shapiro A.
Published: September 2005

The concept of coherence in the field of acoustics apparently means the same thing to all people, but only on an intuitive level, and a formal definition has not been generally agreed upon. The author suggests that the definition used in the theory of multidimensional stochastic processes be adopted by the acousticians. In the present paper, this definition is presented and, when Gaussianity is assumed, an interpretation is given in terms of the relationship between the phases of the signals. The proposed definition leads to the following interpretations: when the processes are highly coherent, the variability of the phase difference between the signals is small; when the processes are incoherent, the phase difference has a high degree of variability. This concept of coherence seems to agree with the intuitive notions currently used in acoustics. (This work was supported by the Office of Naval Research.)
Acoustic Propagation in a Two-Layered Model—Transverse Waves in Bottom
Authors: Eby R. K. , Williams A. O.
Published: September 2005

The complex shear modulus has been measured for samples of Hycar rubber under these conditions: (a) 0.25–2.0 cps at 22°C, with a torsion pendulum; (b) 40 cps–5 kc, between 0° and 41°C, with an electromagnetic transducer; (c) at 10 Mc and 22°C, by McSkimin's method. The “method of reduced variables” allows calculation of the shear modulus, from (b), from 5 cps to 1.2 Mc at 22°C. Measurements (a) and (c) are consistent. From the density and the real part of the modulus, the speed of transverse waves in the rubber has been calculated as a function of acoustic frequency. Results are compared with speeds inferred in a previous paper [R. K. Eby, A. O. Williams, R. P. Ryan, and P. Tamarkin, J. Acoust. Soc. Am. 32, 88 (1960)] that treated acoustic propagation in a model made up of a shallow water layer over a thick slab of this same rubber. The present speeds fall about 25% below those previously inferred, are undoubtedly more reliable, and are still reasonably successful in explaining attenuations within the water layer caused by transverse waves in the bottom, as hypothesized earlier. (A. O. W., Jr., assisted by the Office of Naval Research.)
Iso-Preference Method for Evaluating Speech Transmission Circuits
Authors: Munson W. A. , Karlin J. E.
Published: September 2005

Subjective listening experiments have been conducted to establish a method for determining (1) contours of equal listener preference for a variety of speech transmission conditions in the audio space, and (2) a scale of magnitude of preference between contours based on preference difference thresholds in which the numbers are called TPU's (Transmission Preference Units). The scale is shown to satisfy the transitivity requirement even for widely different transmission conditions. Hence any two conditions with the same TPU number can be predicted to be equal in preference. The method involves only simple AB preference judgments for two brief samples of speech heard consecutively.
On the Design of Underwater Transducer Arrays
Authors: Hanish Sam
Published: September 2005

This paper presents a summary of the state-of-knowledge and some of the unsolved problems facing today's transducer array designers. Of particular importance are the problems associated with the mutual interaction between elements, for element radiating face dimensions smaller than one-half wavelength, when the elements are hard-driven. This aspect of design problems will therefore be the major focal point of the presentation.
Acoustical Design of the Jewett Fine Arts Center—Wellesley College, Wellesley, Massachusetts
Authors: Cavanaugh William J.
Published: September 2005

The music-theater wing of the Mary Cooper Jewett Fine Arts Center at Wellesley College was completed and occupied in the fall of 1958. The building contains a 350-seat auditorium which functions as a music recital hall, lecture hall and, with a portable stage wagon that is moved on tracks from an adjacent drama-workshop, as a small theater. In addition, there are the usual music teaching spaces, including theory classrooms, teaching studios, individual instrumental practice rooms, record-listening rooms and a music library. The room acoustics and sound isolation designs are discussed and the results of some measurements in the completed building are presented. The architects for the building were Paul Rudolph, New Haven, Connecticut, in association with Anderson, Beckwith and Haible of Boston.
Direct and Reverberant Field Statistics in Nearly Hard Rooms
Authors: Lyon Richard H.
Published: September 2005

Calculations of the first-order density of pressure amplitudes in a reverberant room are reported. The random source is presumed to emit a Poisson sequence of pulses which have random amplitudes but identical shapes. It is shown that sufficiently near the source the distribution of pressure amplitudes reproduces that of the source, and that in the reverberant field there is a tendency to normality due to the great image overlap. The conditions for normality are presented, as well as those for statistical independence of the direct and reverberant fields. An experimental illustration of the theoretical ideas is included. Implications for calculations of random fatigue and malfunction are discussed. (Supported in part by Wright Air Development Division and by B+K Instruments, Inc.)
Frequency Distribution of the Normal Modes of Vibration in Rectangular Rooms
Authors: Sepmeyer L. W.
Published: September 2005

Several investigators, notably R. H. Bolt and D. Y. Maa, have examined the statistical distribution of the normal vibration frequencies in rectangular rooms. However, more definitive information would be useful, both for evaluating existing rooms and for optimizing the design of new acoustical laboratories, music rooms or recording studios. Among such information would be more detailed and precise knowledge of the effect of the relative room dimensions on the distribution of its normal modes. An IBM 7090 computer has been programed to determine the frequency distribution of the normal modes of rectangular rooms in each half octave interval over the range of the first four octaves. Frequency has been normalized with respect to the gravest mode, and dimension ratios will be varied from a cube to 5 to 1 parallelepipeds in steps as small as 1%. The principal criterion computed is the rms ratio between the actual mode spacing and the ideal spacing over each half-octave interval. The average mode spacing and, if needed, a more sensitive indicator to help determine the “best” room dimension ratio also will be computed. This paper discusses the equations used, the criteria to be computed, and the results obtained to date.
Optimum Acoustic Criteria of Concert Halls for the Performance of Classical Music
Authors: Winckel Fritz
Published: September 2005

In recent years one has found essential criteria of room acoustics which are responsible for good intelligibility of speech addresses in auditoria. Concert halls which have been built according to these principles are not fully satisfying as to the performance of classical music. The high intelligibility which one deduces from the first reflection pulse and the envelope of the decay curve is not of primary importance for classical music which demands, above all, fullness of tone and blend. An opportunity was afforded to go on tour with the Cleveland Orchestra and to make statistical measurements in 15 different auditoria during their concerts. It has been resulted that the dynamic range attainable by the orchestra is directly dependent on room acoustic qualities. The lower limit of loudness is given by the noise level, variable in the different halls, and the upper limit by diffusion and absorption which are also responsible for the quality or timbre. The behavior of the orchestra is revealed further from the tempo of the music, with the tendency that in the better concert hall with the optimum reverberation time little absorption and good diffusion allow a faster tempo.
Pitch Discrimination for Short Signals as a Function of Interval
Authors: Whitesell Jack C. , Jeffress Lloyd A.
Published: September 2005

Five subjects adjusted the frequency of one oscillator, the “variable,” to match the pitch of a second, the “standard.” The stimuli were presented in sequence, first the standard, then the interstimulus interval, then variable followed by a one-second pause before the next presentation. The subject made his adjustments of the frequency of the variable during the one-second pause. The stimuli were presented binaurally by means of PDR-10 earphones. Five frequencies were used for the standard, 300, 600, 1200, 2400, and 4800 cps, and two durations, 25 and 100 msec. The duration of the variable was kept constant at 100 msec. Two interstimulus intervals were used, 100 and 500 msec. The results of the experiment are in agreement with previous work with respect to the effects of frequency and of duration on the threshold, but show rather more dependence on interstimulus interval than was expected. (This work was supported by the Bureau of Ships.)
Auditory Discrimination of Duration
Authors: Creelman C. Douglas
Published: September 2005

A series of experiments measured human ability to discriminate durations of auditory signals. A two-alternative forced-choice procedure was used: two sine-wave signals of identical amplitude and frequency, differing only in duration, were presented sequentially on each trial. The order of presentation was random, and the task was to state, for each experimental trial, whether the longer signal had occurred first or second. The signals were presented in a background of continuous white masking noise, which was held at a constant level throughout the experiments. In separate experiments the effects on performance of signal voltage, “base” duration, and increment duration were measured. A model was developed within the framework of statistical decision theory to account for the data in terms of a signal-independent “counting mechanism” operating over the relevant durations. Limitations of discrimination are due to uncertainty regarding the end points of the time interval, and memory decay between observations on a single trial. Two parameters were estimated for each observer to fit the data, and the results of two further experiments were predicted. Some implications of the model for psychophysical theory will be discussed. (This research was part of a doctoral dissertation submitted to the University of Michigan. It was conducted under contracts with the Operational Applications Laboratory of the Air Force Command and Control Development Division and the Office of Naval Research. The author is currently a Postdoctoral Fellow at The Johns Hopkins University, Baltimore, Maryland.)
Binaural Lateralization of Cophasic and Antiphasic Clicks
Authors: Flanagan James L. , David E. E. , Watson B. J.
Published: September 2005

Four subjects listened binaurally to equal-amplitude, 100-μsec rectangular pulses. The pulses were delivered at a sensation level of 40 db and at a rate of 10 sec−1. For wide-band reproduction and good transient response, the stimuli were transduced by condenser microphones fitted with ear insert plugs. The relative times of arrival of the pulses at the two ears (both lead and lag) were under continuous control of the subjects. The latter adjusted the interaural time difference to produce fused sound images. The polarities of the pulses were connected for cophasic (rarefaction-rarefaction) and antiphasic (condensation-rarefaction) conditions. Besides the unmasked conditions, lateralizations were made when the two ears were masked by additive, uncorrelated noise, high-pass filtered at 600, 1200 and 2400 cps, respectively. The results indicate that: (a) the interaural time difference for principal fusion of the cophasic condition, both masked and unmasked, is nominally zero; (b) the interaural time difference for principal fusions of the unmasked, antiphasic condition is on the order of 200 to 300 μsec; (c) masking by uncorrelated noise, high-pass filtered at frequencies less than about 1–2 kc/sec, produces antiphasic lateralizations at time differences of approximately ±1/2fc, where fc is the cutoff frequency of the noise; (d) masking by noise high-passed at frequencies greater than 1–2 kc yields lateralizations essentially the same as the unmasked case. Secondary fusions, generally lying off the medial plane, are frequently heard. Earlier calculations on the mathematical models of basilar membrane displacement [J. Acoust. Soc. Am. 32, 1494(A) (1960)] suggest the present results and, in fact, prompted the experiment.
Air-to-Water Transmission Across a Rough Interface
Authors: Scales W. W.
Published: September 2005

The ray solution to the problem of sound transmission across a smooth air-water interface [A. A. Hudimac, J. Acoust. Soc. Am. 29, 916 (1957)] is extended to the case of a rough boundary simulating the ocean surface. A Gaussian distribution of wave slopes was assumed with a variance which is wind dependent. The problem was made two dimensional by considering the wave disturbance limited to swells of indefinite lateral extent. Time averaged transmission losses were computed numerically for a number of source and receiver positions for a point source in air and a receiver in water. A comparison is made for the smooth and rough surface situations. Waves give rise to an improvement in transmission, which increases with wind speed. For a smooth surface the intensity falls off at great ranges as 1/r4, while for the rough surface it tends toward 1/r2. (This work was supported by the Office of Naval Research under contract.)
Underwater Acoustic Ray Analysis
Authors: Graber G. F. , Rubin A. I. , Seymour R. F.
Published: September 2005

A simulation of the trajectory of underwater acoustic rays was implemented on a general purpose electronic analog computer. Equations defining the phenomena are summarized, assumptions are listed and included, and a preferred form of the equations is developed to provide maximum attainable accuracy of solution. Included are considerations of the velocity of sound as an arbitrary function of both depth and range and the effects of reflections from the surface and an arbitrary bottom. Complete computer diagram is presented and special circuits are described. Typical results of depth versus range are shown for varying launch depths and launch angles. A method of obtaining the difference in time required for rays leaving a common emitter at various angles to reach a common receiver is also covered. Finally, provision for computing spreading loss as a function of range is described.
Identification and Discrimination of a Phonemic Contrast Induced by Silent Interval
Authors: Bastian Jarvis , Eimas Peter D. , Liberman Alvin M.
Published: September 2005

At a previous meeting of this Society we described research which indicated that the insertion of silent intervals into recordings of natural productions of certain fricative-vowel sequences is sufficient to induce the perception of stops. For example, introducing a sufficiently long silent interval after /s/ in a production of sore causes the listener to hear store. Further research is presented on the perception of such intervals in the contrast between slit and split. Stimuli containing various durations of silent interval were presented to subjects for (phonemic) identification as slit or split. The same stimuli were presented also for forced-choice discrimination on the basis of any differences the subjects could hear. A comparison of the results indicated that discrimination is most acute in the region of the phoneme boundary. Analysis of the data showed that accuracy of discrimination was determined almost entirely by whether or not the sounds in a given comparison were perceived as different phonemes. Thus, the perception of this acoustic continuum can be regarded as essentially categorical. These results are comparable to those obtained in other studies of consonant perception. The categorical perception of the consonants may be explicable in terms of the categorical nature of their articulatory gestures.
Effects of Vowel Context on the Articulation of Stop Consonants
Authors: Fujimura Osamu
Published: September 2005

Movements of the lips and the mandible were studied by means of a stroboscopic motion picture techinque during the production of intervocalic velar (palatal) and alveolar stop consonants. Three subjects pronounced English words like veto, echo, cookie, okay, etc., in which significantly different labial articulations are required for the first vowel and the second vowel. A film speed of 60 frames/sec was employed for the experiment. Results indicate that during the closure period of nonlabial stops, the labial configuration changes considerably in anticipation of the following vowel. In echo, for example, the lips start to close during the production of the first vowel, and the most conspicuous change in their shape occurs during the closure period of /k/. In general, there is substantial disagreement between the movement of the lips and that of the mandible. Some details of the relation between the articulatory movement and the boundaries found in the sound spectrogram will be given. Implications concerning the acoustic structures of the output sound, such as the “target” of formant transition, will be also discussed. [This work was supported in part by the U. S. Army (Signal Corps); the U. S. Air Force (Office of Scientific Research, Air Research, and Development Command); the U. S. Navy (Office of Naval Research); and the National Science Foundation.]
Methods of Measuring Speech Formant Bandwidths
Authors: Dunn H. K.
Published: September 2005

Accurate measurements of the human vocal tract damping, or the associated formant bandwidths, are not easy. Several different methods have been used by experimenters, with rather widely divergent results. Some of these methods are examined for sources of error, and the conclusions are checked by applying the same methods to the electrical vocal tract. This analog of the human tract can be held constant during a test, can be adjusted in damping, and can be measured accurately by single frequencies. More exact measurement of the real tract would seem to require an artificial source, applied in the pharyngeal cavity, having a lower fundamental and a more accurately known spectrum than the vocal cord tone. A few measurements are reported, obtained by means of an artificial larynx at 30 pulses/sec, applied to the outside of the throat wall. Analysis is made from a tape loop, by means of a narrow band wave analyzer. The results are not wholly satisfactory because the spectrum inside the tract is not accurately known.
Missile Flight Sound Levels—Telemetering Techniques and Interpretation of Data
Authors: Van Houten J. J.
Published: September 2005

Increased emphasis concerning the effects of noise on missile performance has resulted in measurements of flight sound levels utilizing various techniques. In this paper the application of broad band FM telemetering is shown to provide reliable flight measurements. Representative data acquired by use of a condenser microphone system which directly modulated a FM telemetering transmitter is illustrated. Narrow band power spectrum analysis is used to resolve the acoustic signal from system electrical noise. This technique is demonstrated by use of specific examples of telemetered data. Also shown is the correlation of sound level fluctuations and missile maneuvers. It is theorized that these fluctuations result from attached shock wave relocation. Finally, periodic fluctuations in the measured signal strength becomes shown to exist when telemetered signal strength is marginal. These fluctuations result from interference cancellation of the signal when there are two paths to the receiver antenna—one direct path and one reflected off the ground.
Sound Power Measurements on a Competitive Group of Current Room Air Conditioners
Authors: Blazier Warren E.
Published: September 2005

From a practical viewpoint, engineering of “quiet” products must be tempered by the market requirements of size, function and price. It is necessary, therefore, to know something about typical sound levels of particular product types in order to select an objective design criterion for noise control which is not unduly conservative. Such a study has been made on room air conditioners, of a nominal 1-hp capacity, by sampling a total of 12 models from four major manufacturers. Data are presented in terms of the sound power envelope (range of maximum to minimum levels) and presumably represent the current state of art in controlling noise on products of this type. Assuming that at least some of these units have sound levels which are nominally acceptable to the customer, it is surprising to note that a design below NC-50 (relative to a quiet bedroom) is conservative by present standards. The range of data on measurements of the “outside” noise of these units is also included. A form of nomogram is presented for evaluating these results in terms of the non-air-conditioned neighbor as well as pending municipal noise codes.
Instrumentation for Performing Cross-Correlation of Acoustic and Seismic Data on an Analog Computer
Authors: Prout James H. , Crabtree Raymond E.
Published: September 2005

In an experiment designed to study acoustic background noise at high altitudes, a vertical array of two microphones spaced 300 ft apart was hung from a balloon. To determine whether the noise was propagating upward from the jet stream or whether it was caused by local hydrodynamic flow, the signals from the two probes were cross-correlated on an analog computer. The PACE analog computer was chosen for the application because of its ability to handle frequencies as high as 150 cps. An adjustable time delay between channels was provided by a special magnetic tape transport with movable playback heads. Frequency analysis of seismic signals from two large explosions was also performed on the computer by cross-correlating the data with artificially generated sinusoids of various frequencies. Because of the nature of the seismic signal, it is necessary to use special circuitry to permit the accurate measurement of the amplitude of any specified frequency component for any portion of the signal, thus giving an accurate indication of time of occurrence of this particular frequency. Instrumentation for each of these tests is explained and some results are shown for each case. The results of the frequency analysis of the seismic signal are compared with those obtained by conventional filtering methods.
Performance Characteristics of a Wide-Range Mechanical Impedance Pickup
Authors: Bouche R. R.
Published: September 2005

In recent years there has been increased interest in measuring the mechanical impedance at a point on a structure. This paper describes the results of tests conducted during the calibration and evaluation of a mechanical impedance pickup of new design. The pickup weighs 12 lb and includes three force gauge elements and three accelerometers. The calibration results indicate the impedance pickup is suitable for use in the frequency range from ten to several thousand cycles per second. The effective end-mass of the pickup is less than two-tenths pounds and its stiffness greatly exceeds 10 million lb/in. These and other important characteristics indicate the pickup may be used in a wide range of applications requiring mechanical impedance measurements. The test results were obtained on sinusoidal calibration equipment and on a new type shock motion calibrator.
Experimental Study of the Edge Tone Force and the Resulting Acoustic Field
Authors: Unfried Happy Hugh
Published: September 2005

The undulating stream in a jet edgetone system gives rise to a fluctuating pressure distribution over the surface of the wedge, and it is the integrated pressure over this surface which is called the edgetone force. In this series of experiments, the wedge dimensions have been chosen to be small compared to the corresponding acoustic wavelength so that time retardation effects over the surface of the wedge were minimized; hence, it is an effective point load which was measured. The wedge was supported on a pair of zirconate titanate transducers and this system was calibrated by passing a current through the stainless steel wedge placed in a magnetic field. The nozzle block and associated equipment were designed to minimize interference with the acoustic field. In the low-frequency region of operation, measurements show that the acoustic field of the edgetone can be represented by a point dipole whose magnitude corresponds to the measured force. The upper limit for this representation in terms of Reynolds and Strouhal numbers will be discussed. (Taken in part from a M.S. thesis, Dept. of Engr. UCLA, and in part from work supported by the Office of Naval Research.)
Acoustic Radiation Impedance of a Rigid Rectangular Piston in a Moving Airstream
Authors: Plumblee H. E.
Published: September 2005

The effect of air flow on the radiation pattern of a simple source in a moving field, using the “retarded” and “advanced” potential wave equation solutions, is introduced into the classical derivation of radiation impedance. The impedance is derived for two regions of flow velocity: (1) subsonic, where only the “retarded” potential solution is required, and (2) supersonic, where both solutions are necessary. The impedance is derived in terms of the following normalized parameters: piston width/length ratio, (wave number) × (length), and Mach number. A useful application of the results is illustrated through the calculation of acoustic response of a rectangular cavity in a high-velocity airflow. From comparison with measured response, it is concluded that the radiation impedance solutions provide adequate definition of boundary conditions in the actual physical system. (This work has been sponsored by a Wright Air Development Division contract.)
Analysis of Intelligibility of Tissue-Conducted Speech Signals
Authors: Oyer Herbert J.
Published: September 2005

Previous research has revealed that acoustic energy resulting from vocal signals differs in sound pressure level as a function of (1) the anatomical sites at which signal pickup takes place, (2) the composition of the vocal signals, and (3) the fidelity of the recording systems. It has been shown experimentally also that vocal signals picked up out of the external auditory canals of speakers are significantly more intelligible at −15 db and −18 db S/N ratios than are signals simultaneously recorded at the lips. Phonetic analysis of the signals recorded at ear and lips points toward those sounds which cause the ear-recorded signals to be superior. They are the vowels e, ɛ, i. The present research is directed toward a determination of the relative intelligibility of words as they were recorded from six subjects representing two specific body types. Recordings were made at the following anatomical locations: (1) frontal bone of skull, 1 in. above nasalis, (2) mental protuberance of the mandible, (3) immediately below superior thyroid notch of the laryngeal prominence, and (4) at the angle of the mouth. Results of intelligibility testing point toward the use of anatomical sites in addition to the lips for recording of voiced signals. (This research was sponsored by an All-University research grant No. 2771, Michigan State University.)
Automatic Talker Recognition Using Time-Frequency Pattern Matching
Authors: Pruzansky Sandra , Bricker Peter D.
Published: September 2005

A study of machine recognition of talkers has been carried out by digital computer simulation. Test materials consisted of several utterances of the digits one through ten spoken by five male talkers. The talkers were instructed to pause after each utterance, and an appropriate threshold for locating the beginning of each utterance was determined by studying the total energy displayed for each time section. These utterances were converted to time-frequency patterns of spectral energy. The dimensions of these patterns were 17 frequency bands by 60 time sections (1/70 sec each), and the entry at each array point was one of the 1024 possible energy levels. Three-utterance reference patterns were formed by adding together corresponding array points of each of three utterances spoken by the same talker, making five reference patterns for each digit. Talker recognition was accomplished by cross correlating the array of the remaining single utterances of each digit with each of the five reference arrays for that digit and selecting the talker with the maximum correlation. Typical results, for patterns with full bandwidth, indicated a small percentage of errors in talker recognition. Talkers were correctly identified for 94 out of 100 test utterances (two utterances of each digit by each speaker). The errors were distributed throughout the ten digits. Variations in results for patterns which have been converted to a standard duration by a change of time scale and patterns with reduced bandwidth will be discussed.
Speech Sound Classification Based on Signal Statistics
Authors: Bakis R.
Published: September 2005

A method of quantizing the output of a vocoder-type speech analyzer is described. This quantization effectively classifies the sound at each instant into one of a small number (typically, less than 100) of classes, but in such a way as to allow relatively accurate reconstruction of the original unquantized voltages The boundaries of the classes are chosen on the basis of the statistical distribution of the voltages, using redundancies which may exist to reduce the number of classes to a low value. Experimental results are presented, illustrating classification schemes resulting from this procedure. These schemes bear some resemblance to conventional phonetic analyses of speech sounds, but they have the advantage of being derivable by purely mechanical means, without the use of human intuition or judgment. The use of such sound classification for automatic speech recognition is discussed.
Further Progress with Colorless Artificial Reverberation
Authors: Schroeder M. R.
Published: September 2005

In a previous paper [J. Acoust. Soc. Am. 32, 1520 (1960)], artificial reverberators having flat amplitude-frequency responses were described. Such “flat” reverberators preserve the amplitude spectrum of the reverberated sound, thus avoiding any undesirable “coloration.” In the present paper, an alternative approach to colorless artificial reverberation is described in which several “combfilters” (delay lines with multiple reflections) connected in parallel are employed to create a subjectively colorless reverberation. Such reverberators do not have a flat frequency response in the physical sense, but are nevertheless subjectively indistinguishable from flat reverberators if the following (sufficient) conditions are met: (1) The number of combfilters must be at least three. (2) Their delays must be substantially different and incommensurate. (3) The average distance (Δf) between adjacent relative maxima of the frequency response of the reverberator must be smaller than about 5 cps. (This condition is equivalent to requiring that the sum total of the round-trip delays employed in the combfilters exceed 200 msec.) (4) For reverberation times T in excess of 2 sec, the product 〈Δf〉⋅T must be smaller than 10. To meet the additional requirement of a flutter-free transient response, the reverberator must furnish at least one echo per msec for times in excess of about 100 msec after the direct sound. Finally, for the realistic simulation of large halls, it is recommended that a delay of the order of 40 msec be introduced between the direct sound and the reverberation. The advantage of the present “comb-filter approach” to artificial reverberation is that it allows greater flexibility in mixing direct and reverberated sound. Also, the delay lines for the combfilters do not have to meet specifications as stringent as those employed in flat reverberators. A tape recording of artificially reverberated organ music will be played.
Measurement of Probability Density
Authors: Fox Herbert L.
Published: September 2005

The increasing requirements for probability density measurements of the amplitude characteristics of time-varying signals require a careful examination of (1) requirements for probability density analyzers, (2) techniques of measurement, and (3) uncertainty in estimates obtained from samples of finite duration. Requirements discussed include bandwidth and dynamic range, amplitude sampling interval (window width), and necessary readouts. Various techniques of using a probability density analyzer are discussed, with emphasis on the necessity of using different techniques for different types of data and/or use thereof. A method of scanning that guarantees constant uncertainty over the entire range of the probability density is described. An expression for the uncertainty of the measured probability density as a function of sampling time is obtained, and measurements are shown indicating the relationship of sampling time, window width, bandwidth, and probability density to the uncertainty. The normalized variance is shown to be of the form σ2=K/[foTΔxw(x)]. The constant K is derived, using alternate assumptions, and is evaluated experimentally. The analysis in this paper is supported by measurements made with, and on, a commercial probability density analyzer.
Detecting a Signal in Noise: Further Considerations
Authors: Jeffress Lloyd A.
Published: September 2005

In discussing the masking of tonal signals, L. A. Jeffress, H. C. Blodgett, T. T. Sandel, and C. L. Wood [J. Acoust. Soc. Am. 28, 416 (1956)] likened the auditory mechanism for detection to the kind of equipment a physicist might use in attempting to perform the same task. Both involve the use of a narrow filter followed by a detector. The earlier paper described the detector as needing “to be sensitive only to changes in over-all level and to have a certain time characteristic.” The present paper is concerned with properties of the detector, especially its time characteristics for both monaural and binaural stimulation. Some of the problems of signal detection will be illustrated by oscillographic pictures of narrow-band noise and signals for both the homophasic and the antiphasic conditions. (This work was supported by the Bureau of Ships.)
Method of Free Response in Signal Detection: A Simplified Technique
Authors: Egan James P.
Published: September 2005

The “method of free response” refers to the following procedure. In a background of continuous noise, a weak signal is presented several times in a long (2-min) observation interval. The temporal intervals between the occurrence of the signals are randomly selected. No further indication is given to the listener as to the time at which a signal occurs. Over a series of observation intervals, the listener adopts various criteria, and he presses a single “yes-key” whenever he thinks he hears a signal. In our previous studies, the relation between rate of response and time after signal (with the criterion as the parameter) was determined in fine detail. Such an analysis is so tedious (or expensive) that the following simplified technique is suggested. The 2-sec interval just following a signal is compared with a similar interval that occurs considerably after a signal. The proportion of such intervals containing a response is determined for both the near and far intervals, and these proportions are used instead of D (detections) and O (operant level). Only three counter readings are required. The relation between D and O is secured by inducing changes in the criterion of the listener. Estimates of the error introduced by this simplified procedure are presented. (This research was supported by the Operational Applications Office, Air Force Cambridge Research Center.)
Head-Position Identification
Authors: Wright H. N. , Schilling C. W.
Published: September 2005

A binaural recording of traffic sounds that reached an artificial head oriented in five different positions was presented to five subjects, each of whom responded under four different criteria. A Type II ROC curve for the five listeners was linear, P(Y|C)=0.60 P(Y|I)+0.44, and therefore consistent with a guessing model and inconsistent with the detection theory model, Further analysis of the results showed that performance (percent correct and information transmitted) was independent of criterion and did not change systematically over successive trials. (This work was supported by contract from the Office of Vocational Rehabilitation.)
Attenuation of High Frequency Elastic Modes of Propagation in Strips of Polycrystalline Metals
Authors: Meitzler Allen H.
Published: September 2005

Longitudinal and flexural waves in an infinite plate of an isotropic elastic material can be analytically synthesized from a single dilatational potential function and a single rotational potential function. If A is the amplitude of the dilatational potential function and B is the amplitude of the rotational potential function, it is possible to calculate the ratio A/B for a particular mode as a function of frequency once the values of the roots of the appropriate frequency equation are at hand. Calculations of this kind have been carried out for the lowest three longitudinal and lowest three flexural modes and show that, in general, these modes are composite wave motions involving varying combinations of dilatational and rotational components such that A/B varies in amplitude and phase in a given mode as a function of frequency. The primary purpose of this paper is to introduce the “character” of the various modes, defined as [(A/B)(A/B)*], as a concept useful in understanding the selective attenuation of high frequency elastic modes in polycrystalline metal strips. In this type of medium, rotational waves are attenuated appreciably more than dilatational waves; hence a given mode might be expected to exist more strongly over the frequency range in which the total wave motion is predominantly dilatational in character. Experimental data for the attenuation of elastic pulses having 5 to 35 mc carrier frequencies and traveling in strips of polycrystalline aluminum, 5 to 10 mils thick, will be presented. In the results, a good correlation is observed between the frequency range over which a given mode is observed and the frequency range over which, theoretically, the character of the mode is large; i.e., [(A/B) (A/B)*]>1.
On Elastic Constants and Ultrasonic Velocities in Some Crystal Groups
Authors: Mayer Walter G. , Parker Paul M.
Published: September 2005

Expressions are given relating ultrasonic velocities in single crystals to the elastic constants. Without introducing approximations these expressions can be used in a relatively simple manner to evaluate the elastic constants. As an example the procedure is outlined for the trigonal system, based on existing velocity measurements in corundum. Some of the general expressions indicate certain invariance properties of the velocities in simple crystal groups. These properties can be used to check the results of certain velocity measurements or to determine the velocity of certain modes of vibration in some crystallographic directions.
Method for the Pressure Calibration of Pulse Operated Ultrasonic Transducers
Authors: Lester William W.
Published: September 2005

An optical diffraction method for the pressure calibration of pulsed ultrasonic transducers in transparent media is discussed. Results are presented which compare this method with optical calibration by means of continuous waves. As an example of the usefulness of the method, a 5-Mc ceramic transducer of the barium titanate type has been calibrated in water over the temperature range 5–33°C, using 3- to 10-μsec, 5-Mc input signals. Because of the low average input power, stable transducer temperatures are maintained, with no difficulty from streaming or heating of the liquid. (This work was supported by the Office of Naval Research.)
Influence of Different Acoustical Stimuli on the Threshold of the Contralateral Ear
Authors: Loeb M. , Fletcher J. L.
Published: September 2005

A previous attempt to assess the attenuation produced by the intratympanic muscle reflex involved determination of the increase in absolute threshold following contralateral pure tone stimulation. This estimate was considerably smaller than estimates of reflex attenuation obtained by other means. In the present study a 110-db SL sine wave, square wave, narrow band noise, or broad band noise was introduced in one ear and the resultant threshold shift in the contralateral ear was measured. The smallest changes were produced by the square wave and pure tone stimuli. Appreciably larger threshold increases were elicited by the broad band and narrow band noises, especially the latter. Possible explanations of the obtained differences are discussed.
Attenuation by the Acoustic Reflex as Indicated by Contralateral Remote Masking
Authors: Ward W. Dixon
Published: September 2005

One method of measuring the protection afforded by the muscles of the middle ear is to arouse the reflex by means of a loud sound in one ear and measure the concomitant threshold shift in the other. Operationally this amounts to the determination of contralateral remote masking (CRM): “remote” indicates that the arousal stimulus is somewhat higher in frequency than the test tone. When a steady noise is the arousal stimulus, the following phenomena are observed: (1) the CRM adapts steadily, reaching a stable value about 2 min after onset of the noise; (2) a mere change in frequency or level of the arousal stimulus is not sufficient to restore the CRM to a nonadapted level; (3) a 30-sec silent period will restore full CRM after adaptation; (4) maximum protection occurs at 500 cps; (5) medium-frequency octave and crossed-filter bands are somewhat more efficient than higher bands in producing CRM; (6) CRM increases linearly from 85 to 125 db SPL, reaching as high as 40 db in some individuals. Pure tones do not produce a significant sustained CRM. The results are shown to agree with recent animal studies by Wersäll and by Galambos and his associates. (This research was supported by grants B-1122 and B-2199 from NINDB, Public Health Service.)
Temporary Hearing Losses for Bare and Protected Ears as a Function of Exposure Time to Continuous and Firing Noise
Authors: Cohen Alexander
Published: September 2005

Temporary hearing losses for frequencies 250–8000 cps were noted for bare and protected ears (tanker's helmet) following 6-, 12-, and 18-min exposures to recorded 30-cal machine-gun fire and continuous wideband noise of comparable total energy. Threshold losses for both types of noise were generally confined to tones above 1000 cps and tended to become greater with increasing exposure time. Continuous noise caused greater losses than the firing noise under bare ear conditions for the three exposure times. A comparison of these losses against those noted when using the tanker's helmet indicated that the helmet gave significant protection against continuous noise but little protection against the firing noise. A second experiment studied the recovery of 4000 cps threshold losses for a 20-min period following exposure to the noise conditions cited above. Especially for the longer exposure times, rates of threshold recovery from bare ear exposures to continuous noise were slower than those for the firing noise. As compared with the data for bare ears, the tanker's helmet gave faster rates of recovery from losses due to continuous noise but did not facilitate recovery of losses due to firing noise.
Effect of Auditory Fatigue on Differential Intensity Thresholds
Authors: Elliott Donald N. , Riacii Winifred , Silbiger Herman
Published: September 2005

Although the great majority of studies on auditory fatigue have utilized the TTS (temporary threshold shift) as the primary index of fatigue, it also manifests itself in many other ways. Several studies have reported the existence of recruiting in the fatigued ear —as demonstrated at suprathreshold intensities by the use of loudness balance tests, and at threshold intensities by a decrease in the size of the recording attenuator pen excursions when the absolute threshold recovery curve is traced. This paper is a report of the change in the size of the intensity dl's following fatigue. Two parameters have been investigated, viz., the severity of the fatigue as defined by the TTS at the time of the dl determinations, and the sensation level (post stimulation) of the test tone. Both of these factors are found to be related to the change in the dl's.
Age and Sex Differences in Pure-Tone Thresholds. Addendum I: Age Group 51 to 57 Years
Authors: Corso John F.
Published: September 2005

The present paper is an addendum to a laboratory study published in 1959 on the normal thresholds of hearing for pure tones for an age-stratified sample of subjects drawn from a population exposed to minimal levels of industrial noise. It provides audiometric data (250–8000 cps) for 205 subjects in a 51- to 57-year-old group to supplement the earlier data for 500 subjects in four age groups between 18 to 49 years. The results of this study confirm the previously reported findings that (1) there is less than 5 db difference in the average sensitivity between right and left ears, (2) there is a decrease in hearing sensitivity with advancing age for both men and women, and (3) hearing loss with advancing age spreads progressively from higher to lower frequencies. Although women in general have better hearing and less intersubject variability than men, this trend is reversed for the lower frequencies in the 51- to 57-year-old age group. It is concluded that while the onset of presbycusis is later for women than for men, auditory deterioration proceeds at a faster rate in women than in men after about 50 years of age.