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May 1988

Volume 83, Issue S1, pp. S1-S122

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back to top Session G. Psychological Acoustics I and Physiological Acoustics I: Experimental Techniques (Poster Session)
Contributed Papers
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Some research “toys” (A)

Mead C. Killion

J. Acoust. Soc. Am. Volume 83, Issue S1, pp. S15-S15 (1988); (1 page)

Online Publication Date: 13 Aug 2005

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An unbridled curiosity and a certain lack of self‐discipline have resulted in a series of products whose research usefulness sometimes exceeds their immediate economic value to the author's company. On display will be low‐noise, low‐vibration‐sensitivity microphones for measuring spontaneous and stimulated cochlear emissions; a ½‐in in. microphone with switchable diffuse‐field‐inverse filter for simplified KEMAR′ measurements; insert earphones with 70–100 dB of interaural isolation, 30–50 dB of noise exclusion, and various frequency responses (flat pressure response at the eardrum to beyond 12 kHz, flat diffuse‐field‐referenced response, and TDH‐39‐1ike response); and a 1‐mm‐o.d. probe microphone with flat frequency response beyond 10 kHz and two types of precision tubing (soft silicone rubber with approximately 35‐dB wall attenuation and semirigid polyethylene with approximately 50‐dB wall attenuation).
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Hand‐held auditory screener (A)

Lynn S. Alvord

J. Acoust. Soc. Am. Volume 83, Issue S1, pp. S15-S15 (1988); (1 page)

Online Publication Date: 13 Aug 2005

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A new hand‐held auditory screener is presented that utilizes filtered square waves as test signals. Circuit design has resulted in a total size of 2½ × 4 × 2½in., including earphone and cushion. High‐frequency filtering of the square wave signal results in acceptably low levels of harmonic distortion according to ANSI specifications for audiometers [ANSI S3.6‐1970]. Low current drain design provides for a “battery‐low” and “signal‐on” indicator. Individual intensity pots allow for periodic calibration or modification of presentation levels between 20 and 40 dB HL. Initial clinical trials with 50 patients show good agreement with conventional, pure‐tone audiometric screening. Rationale for increased use of hearing screening by health care practitioners is presented.
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Instrumentation for dolphin echolocation experiments (A)

Whitlow W. L. Au

J. Acoust. Soc. Am. Volume 83, Issue S1, pp. S15-S15 (1988); (1 page)

Online Publication Date: 13 Aug 2005

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The use of personal computers has contributed significantly in dolphin echolocation research by providing an inexpensive instrument to measure echolocation signals, monitor and control experimental devices, and to store data. Dolphins typically emit short duration (50–100 μs), broadband click signals with repetition rates that can vary from tens of clicks per second to several hundred clicks per second. In this paper, three electronic measurement devices used with Apple II computers in dolphin echolocation experiments will be discussed. The first device measures the number of clicks emitted, the intervals between clicks, the peak‐to‐peak amplitude of each click, and the time of activation of various switches. The second device measures the frequency spectrum (between 30 and 135 kHz) of each emitted click signal in real time, with a resolution of 15 kHz. The third device is a phantom electronic target simulator that digitizes each emitted click and then retransmits the click signal under control of the computer. The development of these three devices has been a process of evolution in sophistication over a period of several years of dolphin echolocation research conducted at the Naval Ocean Systems Center.
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High‐frequency intensity‐ealibrated threshold measurements using a two‐microphone system (A)

William J. Murphy, Arnold Tubis, and Glenis R. Long

J. Acoust. Soc. Am. Volume 83, Issue S1, pp. S15-S15 (1988); (1 page)

Online Publication Date: 13 Aug 2005

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As a means of calibrating high‐frequency hearing tests without the problems inherent in the use of pressure‐calibrated headphone systems [M. R. Stinson, J. Acoust. Soc. Am. Suppl. 1 87, S75 (1987)], an intensity‐calibrated method of measuring hearing thresholds based on the two‐microphone technique has been developed [W. J. Murphy, A. Tubis, and G. R. Long, J. Acoust. Soc. Am. Suppl. 1 87, S75 (1987)]. The two‐microphone probe is used to estimate the acoustic impedance and the power reflection coefficient of the middle and inner ear system as well as to calibrate psychoacoustic measurements. Various cheeks on the reliability of the long‐term phase calibration are discussed. A computer simulation of the ear canal acoustics is used to determine an appropriate downstream distance of the probe tips from the insert‐earmold sound port. Typical data are given for both objective and psychoacoustic measures obtained with the two‐microphone system. [Work supported by a grant from the Deafness Research Foundation.]
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Modified DigiSound‐16 becomes a precision digital oscillator (A)

W. M. Hartmann, D. L. Edmunds, T. V. Atkinson, and Hal Chamberlain

J. Acoust. Soc. Am. Volume 83, Issue S1, pp. S15-S16 (1988); (2 pages)

Online Publication Date: 13 Aug 2005

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The Digisound‐16 is a two‐channel, 16‐bit DAC‐ADC system with a 32K word buffer. Because of its high performance and low cost it is an attractive signal source for psychoacoustic experiments. The device can be modified to serve as a precision digital oscillator by using the technique of fractional addressing [W. M. Hartmann, J. Acoust. Soc. Am. 82, 1883‐1891 (1987)]. The frequency resolution is 0.003 Hz over the audible range and the total distortion approaches the theoretical limit of − 85 dB for single‐channel operation. The modification requires that the 15‐bit address register be replaced by a 24‐bit increment‐and‐add circuit. Two‐way communication between the modified device and the host computer allows the address increment to be changed continuously so that the oscillator can be manually tuned, as in the method of adjustment. The portion of the buffer that is recycled can be reduced by factors of 2, under program control, to as little as 2K words. The reduced buffer length promotes efficiency while compromising frequency resolution and distortion figures. [Work supported by the National Institutes of Health.]
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Using MIDI, the standard Musical Instrument Digital Interface, for psychoacoustic experiments (A)

H. E. F. Williams, G. L. Gibian, E. N. Harnden, and A. Evans

J. Acoust. Soc. Am. Volume 83, Issue S1, pp. S16-S16 (1988); (1 page)

Online Publication Date: 13 Aug 2005

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MIDI, the music industry's standard Musical Instrument Digital Interface, allows control of commercially available music synthesizers by personal computers, as well as measurement and reproduction of musical performance parameters. The resolution in frequency, amplitude, timbre, and time of the system is adequate for various psychoacoustic experiments. A Yamaha DX7 synthesizer controlled by an IBM PC was used to measure subjects' judgments of the similarity of synthesized tones differing in timbre [G. L. Gibian et al., J. Acoust. Soc. Am. Suppl. 1 82, S68 (1987); G. L. Gibian et al., Audio Engineering Society Preprint #2488 (1987)].
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Demonstration of auditory impedance/reflectance measurement technique, and MIDI enters the research lab (A)

Douglas H. Keefe

J. Acoust. Soc. Am. Volume 83, Issue S1, pp. S16-S16 (1988); (1 page)

Online Publication Date: 13 Aug 2005

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A microcomputer‐based implementation of a two‐microphone wavetube technique for measuring acoustical impedance and energy reflectance will be demonstrated. Data can be fully analyzed with graphical output while the subject is present, thus giving the researcher important flexibility. The Musical Instrument Digital Interface (MIDI) is a standard in electronic music with potential research applications in the areas of acoustical stimulus generation, instrumentation control, and data collection. Selected applications will be demonstrated on the microcomputer with MIDI peripheral devices, and a MIDI software library designed for integration into the research setting will be described. [Work partially supported by NINCDS.]
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A programmable subject response box for psychophysical experimentation (A)

Robert Ling and Edward M. Burns

J. Acoust. Soc. Am. Volume 83, Issue S1, pp. S16-S16 (1988); (1 page)

Online Publication Date: 13 Aug 2005

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Software that utilizes a commercially available touch pad (“Koala Pad”) as an all‐purpose subject response box will be demonstrated. [Work supported by NINCDS.]
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A computer interface for psychophysical and speech research with the Nucleus cochlear implant (A)

Robert V. Shannon, Doug D. Adams, Roger L. Ferrel, Robert L. Palumbo, and Michael Grandgenett

J. Acoust. Soc. Am. Volume 83, Issue S1, pp. S16-S16 (1988); (1 page)

Online Publication Date: 13 Aug 2005

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A custom interface was designed and implemented that, under program control, allows presentation of any available pulse stimulus to a patient fitted with the Nucleus implant. The interface connects to a standard parallel output port of a PC or AT compatible computer. The host computer sends a stream of bytes to the parallel port that specify the configuration of the desired output pulses. Upon receipt of the data, the interface generates the appropriate pulse‐coded sequence to deliver the specified pulses to the patient's external coil. This interface makes it possible to interleave pulses on two or more electrodes, to modulate the amplitude or timing of a pulse sequence, or to sweep a stimulus across the electrode array. This interface allows investigators to conduct psychophysical and speech experiments that cannot be performed with the standard speech processor interface (SPI) normally used to set the patient's device. [Work supported by NIH.]
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IDENTIFY: A program for speech recognition, identification, and categorization experiments on PC compatible computers (A)

Robert V. Shannon, Robert L. Palumbo, and Michael Grandgenett

J. Acoust. Soc. Am. Volume 83, Issue S1, pp. S16-S16 (1988); (1 page)

Online Publication Date: 13 Aug 2005

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IDENTIFY is a program written in Turbo Pascal that accepts any combination of up to 100 speech waveform files as stimuli and up to 25 response categories. The waveform files can be from either ILS or C‐Speech, or any compatible format. The program presents the stimuli in block‐randomized order and waits for the subject to respond on the touch sensitive surface of a small digitizing tablet (Koala Pad). The tablet can be configured to reflect any number and pattern of desired responses. The most common case is when the number of stimuli and responses are the same, producing a standard confusion matrix. In this special case the program outputs an additional data file that conforms to the format for the Speech Information Transmission Analysis (SINFA) [Wang, Behav. Res. Methods Instrum. 8, 471–472 (1976)]. A categorization experiment can be run by specifying many stimuli that span a continuum, allowing only two responses. The program requires a PC or AT compatible computer, a Koala Pad connected to a game port, and a Data Translation DT2801‐A D/A output board. [Work supported by NIH.]
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Transducer and transducer measurement system for the PC (A)

J. B. Allen

J. Acoust. Soc. Am. Volume 83, Issue S1, pp. S16-S16 (1988); (1 page)

Online Publication Date: 13 Aug 2005

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The measurement of several acoustic transducers using a measurement system that is implemented on the PC will be shown. This system uses the ARIEL DSP‐16 and a PC‐6300. The software developed for this purpose, which is called SYSid, was developed first about 8 years ago, and in the past few years was ported to the PC. SYSid will measure the frequency response of a transducer in less than 100 ms with an accuracy that is determined by either the nonlinearities or the noise in the system. Typically, this noise and distortion is more than 80 dB down, giving results that are accurate to 1 part in 104. SYSid also will measure the phase, group delay, impulse response, and the distortion as a function of frequency and level. In the demonstration, an attempt will be made to measure the acoustic impedance of a section of tube, and a rho‐c screen.
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Software for analyzing and plotting scientific data (A)

Walt Jesteadt, Stephen T. Neely, Brian P. Callaghan, and Roger L. Ferrel

J. Acoust. Soc. Am. Volume 83, Issue S1, pp. S16-S16 (1988); (1 page)

Online Publication Date: 13 Aug 2005

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Our PLT program, in use in a number of laboratories, translates a high‐level data plotting command language into Tektronix or HP‐compatible output. It is specifically designed for scientific applications, allowing maximum flexibility in the use of partial log axes, labels, and error bars, with rapid output on a graphics terminal or publication quality hardcopy on a pen plotter or laser printer. GREG, a spreadsheet data analysis program developed in parallel with PLT, allows data to be represented in terms of a five‐dimensional factorial design. It provides convenient data entry from either keyboard or digitizer pad, automatic handling of missing values, data transformations, low‐level statistical analyses, and flexible generation of tables and plots. Plots are generated by producing PLT input files. GREG can call user supplied programs to fit specific models or perform other functions, so the data analysis framework can be easily expanded. Both PLT and GREG run under RT‐11, TSX, UNIX, and MS‐DOS; they will be demonstrated on a portable AT clone. [Work supported by NIH.]
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Application of digital signal processing techniques to research on sound localization (A)

Frederic Wightman and Doris Kistler

J. Acoust. Soc. Am. Volume 83, Issue S1, pp. S16-S17 (1988); (2 pages)

Online Publication Date: 13 Aug 2005

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The availability of fast, general‐purpose laboratory computers has dramatically changed the way research is done in many areas of psychoacoustics. In studies of sound localization, for example, standard digital signal‐processing techniques are used to synthesize stimuli for headphone presentation that contain all of the localization cues (e.g., interaural differences, pinna effects) that are available from sounds presented in free field. This brings to localization research a much needed degree of stimulus control and specificity, and allows us to address certain basic issues that were heretofore inaccessible. This presentation will survey the general principles involved in the application of digital signal processing to localization research, and will discuss the practical limitations of the specific techniques that are used (e.g., FFTs, FIR filters, inverse filtering). Empirical data will be shown in order to address questions relating to the classic tradeoffs among signal/noise ratio, bandwidth, word‐length, and sampling frequency. The feasibility of real‐time digital processing in currently available PC‐based systems will also be discussed. [Work supported by NIH, NSF, and NASA.]
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Classical respiratory conditioning used in auditory psychophysics of the goldfish: Methods and results in detection and discrimination studies (A)

R. Fay, S. Coombs, and C. Wheeles

J. Acoust. Soc. Am. Volume 83, Issue S1, pp. S17-S17 (1988); (1 page)

Online Publication Date: 13 Aug 2005

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Classical conditioning of respiration has been used to study many aspects of hearing in the goldfish, including absolute and masked thresholds, frequency, intensity, and time discrimination thresholds. A restrained fish is presented with a 7‐s auditory signal that ends with a brief electric shock across the body. Shock causes an unconditioned suppression of respiration lasting several seconds. Several pairings of the auditory signal with the shock results in respiratory suppression during the signal. Initial conditioning is rapid (conditioned responses appear within the first 10 to 20 trials), several thresholds can be obtained in one day, and thresholds can be obtained for individual animals for several years. Critical factors for success with this method include the measurement of respiration (using a thermistor), method of animal restraint, levels of electric shock, intertrial intervals, false alarm estimation, overall respiratory rate, subject selection, and water conditioning. Details of the methods and procedures that have been found useful in conditioning and threshold definition will be given along with illustrative data on threshold values and stability in masking and intensity discrimination experiments.
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Psychoacoustical measurements with delays in neonates' vocalizations (A)

Lincoln Gray

J. Acoust. Soc. Am. Volume 83, Issue S1, pp. S17-S17 (1988); (1 page)

Online Publication Date: 13 Aug 2005

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Newborn chickens momentarily delay their usually incessant peeping when they hear either an onset or change in an auditory stimulus. This unconditioned response provides several ways to study the early development of hearing. (1) Receiver operating characteristics, drawn from pooled histograms of responses on stimulus and control trials, strongly suggest that this response is a measure of auditory detection. Areas under these curves can be used to quantify the auditory abilities of newborn subjects. (2) Adaptive procedures based on paired comparisons of stimulus and control trials provide rapid estimates of thresholds and difference limens. A five‐frequency audiogram, for example, can be obtained from a newborn chick in several minutes. (3) Responses to all possible pairs of transitions between multiple stimuli can be used as proximity data for multidimensional scaling algorithms. This provides a “map” of how auditory perceptions change immediately after birth. (4) Neonates' responses to naturalistic stimuli may be different than those to more arbitrary puretone and noise signals. Delayed vocalizations also occur in other animals, allowing comparative studies of auditory development. [Work supported by NIH.]
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Techniques for free‐field testing of spatial attributes of acoustic signals (A)

R. Wayne Gatehouse

J. Acoust. Soc. Am. Volume 83, Issue S1, pp. S17-S17 (1988); (1 page)

Online Publication Date: 13 Aug 2005

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For a number of years Gatehouse and his coauthors have reported on studies of various aspects of spatial acoustics (localization, depth perception, masking) done under free‐field conditions of signal presentations. Basically, the paradigms have involved comparisons of the responses of normally hearing, artificially degraded (monaural or binaural occlusion by ear muffs and plugs), and real hearing impaired subjects of various types and degrees of loss, under conditions that presumably mirror the more normal hearing environment, i.e., reverberant or semireverberant rooms, all types of signals (noises, tones, speech), from positions that vary in both azimuth and elevation and from all around the subjects. In this session, some of these techniques and conditions will be displayed in a more detailed manner than is usually available in verbal presentations of the methodologies. [Work has been supported under various NSERC and MRC grants.]
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The use of multidimensional scaling techniques for revealing perceptual categories for complex stimuli in animals (A)

Robert J. Dooling, Kazuo Okanoya, Susan D. Brown, and Thomas J. Park

J. Acoust. Soc. Am. Volume 83, Issue S1, pp. S17-S17 (1988); (1 page)

Online Publication Date: 13 Aug 2005

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A combination of operant conditioning and multidimensional scaling techniques for demonstrating natural perceptual categories for complex sounds in small birds are described. Birds are trained using operant conditioning procedures on either a same/different discrimination task or on a task requiring the detection of change in a repeating background. Response latencies are used to construct similarity matrices, and multidimensional scaling procedures are then used to produce spatial maps of complex sounds reflecting perceptual organization. Stimulus similarity is represented by spatial proximity. Stimulus groupings in multidimensional space indicate perceptual categories that can be confirmed by cluster analyses. These procedures have been used to study the perception of complex sounds such as bird calls and speech in small birds, but these techniques should also be useful in examining the perception of complex sounds in other animals. [Work supported by NIH.]
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Small‐sample statistical analysis of Levitt's psychophysical procedure (A)

Brent W. Edwards and Gregory H. Wakefield

J. Acoust. Soc. Am. Volume 83, Issue S1, pp. S17-S17 (1988); (1 page)

Online Publication Date: 13 Aug 2005

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Levitt's 2IFC adaptive procedure is a widely used psychophysical method for estimating detection or discrimination threshold. The commonly accepted analysis shows that the average of the levels at which reversals occur approaches the 0.707 point on the psychometric function when the levels are updated according to the “2‐up, 1‐down” rule. This analysis is based on a continuous representation of the psychometric function and on the asymptotic behavior of the tracks. The more practical case was considered in which the psychometric function is represented by a set of discrete levels and in which the stopping criterion is a fixed number of reversals. A Markov model for the reversal levels is proposed as a method for determining the statistics of the estimator. Based on this approach, the estimator is significantly biased for small numbers of reversals, and decreases monotonically as this number becomes large. Convergence to the 0.707 point, however, is not guaranteed and depends on the sampling of the psychometric function. These results hold, in general, regardless of the form of the psychometric function or the method by which the psychometric function is sampled. More rapid convergence with less bias may be achieved by tailoring the sampling to the psychometric function. Several approaches to the design of such sampling will be discussed. [Research supported by AFOSR.]
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Positron emission tomography (PET) as a technique for studying patterns of regional cerebral function associated with auditory perception and speech production (A)

John J. Sidtis, Vijay Dhawan, James R. Moeller, Stephen C. Strother, David Eidelberg, and David A. Rottenberg

J. Acoust. Soc. Am. Volume 83, Issue S1, pp. S17-S17 (1988); (1 page)

Online Publication Date: 13 Aug 2005

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PET is an in vivo technique for the measurement of patterns of regional cerebral blood flow (rCBF), metabolic rate for glucose (rCMRGlu), and the uptake of neurotransmitters such as dopamine. Scanning times range from 1 min for rCBF, to 2 h for dopamine, with rCMRGlu scans requiring 1 h. Studies of rCMRGlu in patients with movement disorders and dysarthria have demonstrated correlations between voice onset time abnormalities and rCMRGlu in the basal ganglia; similar studies with dopamine are currently in progress. Because of the shorter scan time, rCBF studies are better suited for perceptual studies. Using a steady‐state C0150 inhalation technique, 14 1‐min scans can be obtained back‐to‐back. The C0150/PET has been used to study rCBF patterns in within‐subject studies that include both stimulated and unstimulated periods, with stimuli such as broadband white noise, music, and complex tones in a pitch discrimination task. Results to date demonstrate that both regional and global changes are associated with stimulation states. PET provides a unique technique for studying human cerebral physiology associated with auditory perception and speech production. [Work supported by NIH.]
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Acetylcholine effects on tone elicited single unit responses in rat auditory cortex (A)

H. K. Rucker

J. Acoust. Soc. Am. Volume 83, Issue S1, pp. S18-S18 (1988); (1 page)

Online Publication Date: 13 Aug 2005

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Urethane (1.5 g/kg) anesthetized male Sprague‐Dawley albino rats were affixed with a head bolt and a burr hole placed over the temporal cortex and wound edges infiltrated with lidocaine before being placed in an electrically shielded, sound attenuating chamber. Tone bursts were generated with a software controlled M308 stimulus generator and 300 stimulus control module (Modular Instruments, Inc.) and sent via a secondary amplifier to a Realistic 40‐1377 loudspeaker. Sound‐pressure levels were determined with a calibrated ¼‐in. B & K 4135 condensor microphone. Single unit activity was recorded via the center barrel (4M NaCl) of glass multi‐pipette. Drug barrels contained acetylcholine chloride (0.5M, pH 4.5), atropine sulfate (0.02M, pH 5.0), and glutamic acid (monosodium salt, 0.2M, pH 8.0). Sixty‐six percent (n = 21) of units characterized demonstrated an atropine sensitive Ach facilitated response to near best frequency tone bursts. An atropine insensitive Ach induced response decline was noted in 13% (n = 4). [Work supported by NSF Grant BNS 8617937.]
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A new set of auditory demonstrations (A)

A. J. M. Houtsma, T. D. Rossing, and W. M. Wagenaars

J. Acoust. Soc. Am. Volume 83, Issue S1, pp. S18-S18 (1988); (1 page)

Online Publication Date: 13 Aug 2005

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A new series of auditory demonstrations for classroom use, modeled after the well‐known “Harvard Tapes” [D. M. Green, Harvard Laboratory of Psychophysics (1978)] was developed and produced. The demonstrations are issued on a compact disc (CD), are accompanied by a 92‐page explanatory booklet, and fit conveniently in a handy slide box. Almost all demonstrations were synthesized digitally on DEC Vax 11/780 and MicroVax II computers, converted to analog signals with DSC‐200 16‐bit two‐channel D/A converters, and recorded on 16‐bit PCM video tape. Brief spoken introductions to each demonstration were recorded on similar PCM video tape. Editing of the master tape was done digitally in 1630 format on ¾‐in. U‐Matic tape, from which the CD master was made. The CD medium, allowing a S/N ratio of up to 90 dB, is very appropriate for sound demonstrations that require a clean acoustic background. The medium also provides random access to individual demonstrations or parts thereof, allows preprogramming of any desired combination of demonstrations, and is much more resistant against wear and tear than traditional tapes or records. The CD can be played on any CD player capable of handling selections from up to 80 tracks. The demo set is available through the ASA.
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An auditory filterbank design and its hardware implementation with DSP (A)

Takashi Komakine and Tatsuya Hirahara

J. Acoust. Soc. Am. Volume 83, Issue S1, pp. S18-S18 (1988); (1 page)

Online Publication Date: 13 Aug 2005

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As the first step in constructing a signal processor to simulate the human auditory function, a computational filterbank with cochlear frequency analysis function was designed [T. Komakine and T. Hirahara, Tech. Rep. IEICE Jpn. SP87‐45, 65–72 (1987)(in Japanese)] and its hardware with DSP implementation completed. From the various cochlear filtering models proposed to date, a cascade/parallel type filterbank model [e.g., R. F. Lyon, ICASSP82, 1282–1285 (1982)] was chosen followed by design of a filterbank composed of 61‐channel IIR filter blocks. Each filter block was spaced at ⅓ Bark intervals from 1 to 21 Bark. Parameters for each filter were decided by taking into account observed physiological data and filter stability. The parameter optimizing effects of the filter were discussed. Sound spectrograms produced by using the filterbank showed that this auditory filterbank model allows excellent frequency and time resolution. A hardware implementation of the filterbank, composed of floating‐point DSP chips and a host controller, was studied. In addition to the computational filtering functions, some control functions, such as automatic addressing previously done in the host controller, are distributed to each DSP. Therefore, control of each DSP unit by the host is easier. More sophisticated functions are easily added. This is in part due to a program memory commonly accessible both from each DSP as well as the host. Finally, the auditory filterbank is capable of a frequency analysis of up to 3000 channels within 5 ms, which is equal to the time required for the wave to travel the basilar membrane. It is also expandable to allow hardware implementation with more than 30 000 cascaded DSP units.
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Whistling with heavy gases or at elevated pressures (A)

R. Stuart Mackay

J. Acoust. Soc. Am. Volume 83, Issue S1, pp. S18-S18 (1988); (1 page)

Online Publication Date: 13 Aug 2005

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Most divers can speak easily but cannot whistle at elevated pressure. Suggestively, sperm whales are not known to whistle. However, rigid flutes or bird calls remain easy to blow “at depth”; frequency changes little. The author can just whistle tunes at 4 atm (equivalent water depth of 30 m). At one atmosphere he could whistle with a breath of half air and half sulfur hexafluoride but not with pure SF6 (density relative to air of 5.1); alternate whistling in and out aids the observation. Possibly this is due to increasing gas density yielding damping vibrations in the soft tissues shedding vortices. Caution: Limiting duration prevents suffocation; possible impurities in SF6 are toxic and corrode metal. The SF6 lowers whisper and speech tones but not hum and whistle frequencies; reflex changes in the throat can confuse the latter. A colleague, Jon Pegg, can whistle tunes normally as well as through his nose with mouth closed and vocal cords constricted to a slit. His nose whistle quenches at 20 m and the normal one at 10 m, suggesting sites of different stiffness. It is said to be impossible for organ pipes to sound if the stream is denser than the surroundings. Several blew well (frequency lowered) with octafluoropropane (density 6.5) in air.
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Digital synthesis of binaural auditory localization azimuth cues using headphones (A)

Richard L. McKinley and Mark A. Ericson

J. Acoust. Soc. Am. Volume 83, Issue S1, pp. S18-S18 (1988); (1 page)

Online Publication Date: 13 Aug 2005

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A laboratory demonstration prototype of a digital auditory localization cue synthesizer has been developed. This synthesizer uses a single audio input that is separately processed in real‐time for independent presentation to each ear using headphones. The headphone presented acoustic signals are easy to localize and appear to be out of head. The acoustic image is stabilized for head movement by use of a three‐space head tracking device. The paper will describe the salient parameters of the design. A description of the psychoacoustic and electroacoustic measurements that led to the design will be presented. Human performance data on free field, simulated, and synthesized localization cues will be described and a real‐time interactive demonstration will be available for interested listeners.
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