bluuebunny/crossref_metadata_2025 huggingface dataset:
Wide-hand noise reduction of noisy speech
Authors: Magill D. T. , Un C. K.
Published: October 2005

We present a new technique that reduces wide-band noise in noisy speech. The concept of the approach is based on residual or voice excited linear predictive coding (LPC). The technique can be used in either a digital or an analog communication link that is degraded by quantization or receiver thermal noise, or both. The technique capitalizes on two effects. (1) Linear predictive coefficients representing an all-pole vocal tract transfer function can be calculated unaffected by noise, if the noise present in speech is additive and white. (2) The signal-to-noise ratio of voiced sound in the fundamental frequency region of the voice band is much higher than that in the high-frequency region. The basic procedure for conditioning noisy speech is as follows. When a voice signal that has been distorted because of quantization or thermal noise, or both, is received at a receiver, the decoded signal is analyzed to determine a set of LPC coefficients by the autocorrelation approach. The noise affects only the diagonal elements of the autocorrelation matrix, since it is not correlated with speech. Hence, to find the coefficients that are unaffected by noise, the noise power is subtracted from the diagonal elements of the matrix. The power of white noise is measured from a speech pause. The residual signal or noisy speech is low-pass filtered at a cutoff frequency of 600 Hz and then passed through a nonlinear distortion device to broaden its bandwidth, This nonlinearly processed signal is used as the input to excite the filter which has been formed by the LPC coefficients. Initial testing of the conditioner by computer simulation has yielded promising results.
Piecewise linear quantization of linear prediction coefficients
Authors: Un C. K. , Yang S. C.
Published: October 2005

We present a new coefficient-coding method of linear predictive coding (LPC) of speech. It employs piecewise linear quantization and requires statistical properties of the LPC reflection coefficient {ki}. In coding the coefficients we first construct histograms of the distribution of each coefficient values obtained from analysis of many different speakers. Then, rather than making quantization levels uniform throughout the range of coefficient values, we divide the range of each coefficient or its transformed version into three regions and assign a different number of quantization levels to each region. Before the range of each coefficient is divided, the minimum value R0 and the maximum value R3 of each ki are set on the basis of experimental observations. Division of each range into three regions is made by specifying R1 and R2 according to the distribution of coefficient values and desired quantization accuracy in each region. Although the piecewise linear coding places its importance on the density of frequencies of the coefficient values, it does not neglect the importance of the spectral sensitivity aspect. In our informal subjective listening test it was observed that the quality of synthetic speech with the transmission rate of 2.4 kbits/sec coded by piecewise quantization was equivalent to the quality with the rate of 3 kbits/sec coded by linear quantization. The piecewise linear quantization scheme can also be used for the arc sine and log area ratio coding methods that normally employ linear quantization. In this case one can expect approximately the same amount of saving of transmission rate as for the nontransformed reflection coefficients.
Best recursire fit: Some experiments on synthetic data and on real speech
Authors: Neuburg Edward P.
Published: October 2005

A synthetic signal was formed by passing a burst of energy through an all-pole digital filter, and Gaussian noise was added to the output. (This is the model of speech production that best recursive fit (BRF) was designed to analyze.) Both linear predictive coding (LPC) and BRF were used to try to recover the filter parameters; at all significant levels of noise BRF is an order of magnitude more effective than LPC. A similar experiment was done on real speech; both LPC and BRF were done (pitch synchronously) on voiced speech, then noise was added to the speech and LPC and BRF were done again. Recovered pole frequencies and bandwidths vary as noise is added; the expected “error” is less for BRF than for LPC. BRF analysis and resynthesis of speech will be demonstrated by the playing of a short passage.
Adaptive linear prediction filtering for speech analysis
Authors: Heisey D. L.
Published: October 2005

An adaptive linear prediction filter utilizing Widrow's least-mean-square noisy gradient algorithm is under study as part of TRW's ongoing voice processing effort. The computational simplicity of the algorithm, as well as its ability to quickly adapt to changes in the statistics of the input speech, makes the algorithm attractive for voice processing. Unlike other block form LPC algorithms which process the data in frames at fixed update rates (autocorrelation and covariance methods), the filter coefficients are updated at every data sample and a new set of LPC parameters generated at any desired time (variable frame rate). Previous work with the adaptive filter has focused on a cascaded realization for formant analysis [L. B. Jackson and J. Bertrand, Proc. IEEE International Conference on Acoust. Speech Signal Process., pp. 84–86, April, 1976] or input signals other than voiced speech [L. J. Griffiths, IEEE Trans. Acoust. Speech Signal Process. ASSP-23, 207–222 (1975)]. Preliminary results suggest that the original algorithm as implemented by Griffiths may be unsuitable due to the presence of pitch pulses in voiced speech. A modified version of the algorithm using a fixed step size per iteration appears to minimize this problem resulting in better spectral estimates. Numerical examples which illustrate the properties of the algorithm as compared to other LPC algorithms currently used in speech analysis are included.
The phoneme vocoder: Speech at 160 pps and below
Authors: Gillmann Richard
Published: October 2005

A computer-based system has been developed which transmits speech at extremely low bit rates, using binary-coded sequences of phonemes. This phoneme vocoder translates digitized speech into phoneme strings using a system developed originally for an automatic speech understanding system. The phoneme strings are then encoded and transmitted to a digital speech synthesizer for reconstruction. Compression to 160 bps has been achieved with the current system. By encoding the phonemes into syllables, rates as low as 38 bps appear to be possible. Results of a study of the accuracy of the phoneme vocoder at the word level will be presented, and a tape recording will be played.
On the design of sub-band coders for low bit rate speech communications
Authors: Crochiere R. E.
Published: October 2005

Digital waveform coding in sub-bands offers attractive possibilities for economically encoding speech at bit rates in the range of 7.2–16 kbits/sec. The design of sub-band coders involves the consideration of a large number of parameters and “tradeoffs.” The number of sub-bands, the partitioning of sub-bands (and gaps between sub-bands), allocation of bits among sub-bands and compromises between bits/sample and bandwidth must be considered. In addition, constraints are introduced by practical considerations of integer-band sampling, multiplexing of the digitized sub-band signals, and synchronizing and framing of the data. Design guidelines, based on extensive computer simulations and subjective comparisons, are presented for choosing these parameters and a method for synchronous multiplexing of the sub-band data, without buffering, is proposed. Several examples of sub-band coders for transmission rates of 7.2, 9.6, and 16 Kbits/sec are presented and compared against that of ADPCM and ADM coders. Tapes will be played to demonstrate the quality of the sub-band coders.
Subjective effects of anomalies in packetized speech transmission
Authors: Forgie J. W.
Published: October 2005

When speech is transmitted in a packet-switched network the variability in packet delays inherent in such a net tends to produce occasional anomalies or “glitches” in the output speech when packets fail to arrive at the destination in a timely fashion. While the frequency of occurrence of these anomalies can be minimized at the expense of buffering and increased overall speech delay, it is likely that a practical network design would represent a compromise which allowed some degradation of the output speech under worst case load conditions. To provide some basic data on the subjective effects of such anomalies a series of listening and communicability tests has been undertaken using speech in which anomaly effects have been introduced by simulation techniques. Anomalies resulting from packet losses due to delay dispersion as well as variation in average delay are introduced at controlled rates in simulated networks using a variety of speech encoding techniques and packetization strategies. Preliminary test results will be presented and a tape with examples of some anomaly effects will be played. [This work was sponsored by the Defense Communications Agency for the Department of Defense.]
Acoustical design of a major international airport
Authors: Nugent R. E. , Salter C. M. , Freytag J. C.
Published: October 2005

The application of noise-control engineering during the design of a new international airport for a future population center presents unique challenges. This airport will be constructed on a site 25 km long and 12 km wide and is 22 km from any populated area. The airport facilities include two international terminals, two domestic terminals, a large mosque, a control tower, a royal facility, and many support buildings. This paper discusses the organization and procedures used to provide noise-control engineering support to the engineering and architectural firms designing these various facilities. Included is the methodology used in determining the acoustical criteria, the sound isolation and speech privacy requirements for the various spaces. Sound isolation details and unit cost comparisons are presented.
Recent experience with the California Insulation Standards
Authors: Hilliard John K. , Bricken Gordon L.
Published: October 2005

The California Noise Insulation Standards have been in effect for about two years. This unique piece of legislation has for the first time placed minimum requirements for noise insulation within all building codes in the state. Many such modifications of the building code takes time to filter down to the local governments. Lacking the acoustical expertise, local governments have varying responses to the Noise Insulation Code and varying approaches to responsibility for imposing requirements. One result of the rode has been stricter reviews of current practices in constructing party walls and floor-ceiling assemblies in the field. This has revealed many difficulties in current designs and most particularly in practices in the building industry for the installation of such assemblies. This paper will address the experience of Bio-Acoustical Engineering Corp. as a consultant to builders and local building officials on this issue in over 500 different separate reports in the last two years. Some recommendations on the implementation of the code will be included.
Music, speech, and hearing: A general education course in descriptive acoustics
Authors: Strong W. J. , Palmer E. P. , Dudley J. D.
Published: October 2005

A general education course in descriptive acoustics for students in music, speech, and related areas is described. Resource materials include books, study guide, films, audio tapes, demonstrations, and walk-in labs. The course has been evolving as it has been offered over the past 16 semesters. However, our university has recently adopted a general education program which allows students to test out on a course without formally enrolling. This has necessitated making the exams and resource material more generally available, and poses a particular challenge for lab-oriented activity. Much of this paper is concerned with aspects of a walk-in hands-on lab designed to meet these new requirements.
Teaching psychoacoustics to a mixed audience of music, science, and humanities majors
Authors: Roederer Juan G.
Published: October 2005

Boundary conditions imposed by widely different student backgrounds in an elementary course on musical psychoacoustics demand careful identification of relevant topics and formulation of questions that are attractive and comprehensible to all sectors involved. [J. G. Roederer, Introduction to the Physics and Psychophysics of Music, 2nd ed. (Springer, New York, 1975)]. The following topics and related questions have been found suitable for treatment in such a course: (a) Perception of spectral pitch of sinusoidal tones (Why do we have good pitch resolution in spite of poor cochlear frequency analysis?), (2) Perception of subjective pitch of complex tones (Why do we perceive tones made up of many harmonics as single-pitched wholes? Why do we perceive a missing fundamental?), (3) Consonance, dissonance, and tonic dominance (Why do we call all notes that are octaves apart by the same name? Is harmony the result of a learning process?), and (4) Hemispheric specialization of speech and music processing in the brain (Which tasks of music perception involve preferentially one cerebral hemisphere? To what extent do language centers of the brain become involved in musical information processing?) The talk will focus on the evaluation of teaching experience in class and laboratory covering the above topics.
The remotely operated multiple array acoustic range (ROMAAR) for the direct measurement of airplane flyover noise footprints
Authors: Hilton D. A. , Henderson H. R.
Published: October 2005

The ROMAAR now in operation at NASA will allow direct measurement and display of aircraft noise contours in several measurement units during takeoff, landing, and flyby operations. Direct measurement of contours allows determination of the statistical variation of contours due to the atmosphere or aircraft operational parameters, a measure of the impact of various noise reduction techniques and hardware on ground noise footprints, and the ability to supply direct inputs for improvement of the NASA Aircraft Noise Prediction Office predictive program. The methods, techniques, and equipment developed for the ROMAAR concept are applicable to CTOL, STOL, General Aviation, and VTOL aircraft. ROMAAR represents a unique combination of state of the art digital and analog noise recording methods, computer-controlled digital communications methods, radar-tracking facilities, quick look weather capabilities, and a large data handling facility complemented by a large capacity curve fitting and plotting routine. The ROMAAR is set apart from the standard airport noise monitoring system by having the unique features mentioned above plus the fact that up to 33 separate (but simultaneous) noise measurements are made for each aircraft overflight. This paper will present a description of the ROMAAR facility and its uses along with actual measured and predicted noise footprints.
Measuring atmospheric temperature and humidity for aircraft flyover noise testing
Authors: Hogstedt C. R.
Published: October 2005

The measared of near-surface and upper-air temperature and humidity is required during flyover noise tests to adjust the noise data to reference-weather conditions or to classify the data with respect to various criteria. Parallel advancements in equipment, test techniques, and noise data adjustments have occurred in the last several years at the Douglas Aircraft Company. Measurements of near-surface temperature and humidity began with a psychrometer hand-held at four feet above the ground and subsequently evolved through a series of instrument types mounted 33 feet above the ground on a trailered tower. Upper-air (or sound-path) weather data were initially acquired from local weather bureaus and later from various instrumented light aircraft. The more notable improvements are the use of dew-point sensors for humidity measurements, increased data sample rates, and system performance monitoring. Current instruments, along with sufficient attention to time and space dependencies, appear to furnish an adequate definition of the test atmosphere. However, more data are required to quantify and evaluate the actual accuracy of the system.
Source array for generating higher order acoustic modes in circular ducts
Authors: Wyerman B. R. , Reethof G.
Published: October 2005

A unique source array has been developed for the generation of both spinning and nonspinning higher-order modes in a circular duct. The array consists of two concentric rings of sources. Through individual control of the response of each element, the array provided phase and amplitude control in the radial as well as circumferential directions. Radial modes shapes were measured in a 12-in. diameter anechoically—terminated hollow duct. These modes could be generated at their cut-off frequency and throughout a frequency range extending to the cut-off frequency for the next higher-order radial mode. Comparisons are given between theory and experiment for the generation of specific modes. The radial dependence of the measured mode shapes was enhanced considerably by the design of this array. The results indicate a significant improvement over previous mode generation mechanisms. The contamination of the generated mode by additional spurious modes is also considered for variations between individual elements within the source array. [Work supported by NASA Langley Research Center.]
Sound attenuation in acoustically lined curved ducts in the absence of fluid flow
Authors: Ko Sung-Hwan , Ho Louis T.
Published: October 2005

A study has been made of the sound attenuation in a lined curved duct with rectangular cross section. In this study, the derivation of eigenvalue equation was based on the continuity of the normal component of the particle displacement and the matching of the acoustic pressure on the acoustic lining surface. The sound attenuation was calculated using the acoustic energy expression for the waves propagating in a curve duct. For a given duct geometry and known acoustic lining impedances, a computer program was developed to solve for the eigenvalues and to obtain the sound attenuation of the propagating waves in the lined curved duct. It was found that the fundamental mode was least attenuated. The total sound attenuation was calculated assuming that the amplitudes for all propagating waves were equal at a given frequency. Effects of aspect ratio, bent angle, and the acoustic impedance on the sound attenuation were investigated in the present work. [Work sponsored in part by Naval Sea Systems Command Code 037.]
Attenuation of jet noise in a lined augmenter tube: A model study
Authors: Vér István L. , Andersen Douglas W. , Grunnet James L.
Published: October 2005

To obtain needed information for the design of an air-cooled jet aircraft run up noise suppressor system, scale-model investigations have been carried out. Acoustical, thermal, and pumping performances of various augmenter tube configurations, utilizing a jet exiting from a 2.5-in.-diam. nozzle as a function of pressure ratio (up to 2) and temperature (up to 3300°F), have been investigated. Base data, obtained with hard-walled augmenters, have been already described in an earlier paper [I. L. Vér, D. W. Andersen, and J. L. Grunnet, J. Acoust. Soc. Am. 59, S95(A) (1976)]. This paper reports on the measured attenuation of the jet noise in lined augmenter tubes of various constructions as functions of jet temperature, pressure ratio, augmenter tube length, and frequency. The effects of noise regeneration at the exit plane, pumping performance, and the temperature and flow velocity distribution will also be discussed. [This work was supported, in part, by the U. S. Navy, Southern Division, Naval Facilities Engineering Command.]
Leaking modes in the propagation of atmospheric acoustic-gravity waves
Authors: Kinney W. A. , Pierce A. D.
Published: October 2005

Attempts to predict the transient acoustic pressure pulse at long horizontal distances from large explosions in the atmosphere have adopted a model atmosphere bounded above by a halfspace of finite sound speed and have represented the waveform as a superposition of contributions from dispersively propagating guided modes. Certain modes at low-frequency decay exponentially (leaking modes) with increasing distance of propagation. Until now, the practice has been to neglect the contributions from leaking modes at low frequency. The theory associated with leaking modes at low frequency. The theory associated with leaking modes has been previously discussed in detail [V. A. Kinney, A. D. Pierce, and C. Y. Kapper, J. Acoust. Soc. Am. 58, S108(A) (1975)]. Examples of numerically synthesized transient waveforms are now exhibited with and without the inclusion of leaking modes. The inclusion of leaking modes results in waveforms each having a marked beginning rather than a low-frequency oscillating precursor of gradually increasing amplitude as previously obtained. Also, the revised computations indicate that waveforms invariably begin with a pressure rise; a result supported by other theoretical considerations and experimental data. [Work supported by Air Force Geophysics Laboratory.]
Acoustic/seismic coupling phenomenology
Authors: Constantine J. G. , WaLker R. B.
Published: October 2005

This paper summarizes the results of a recent experiment that was designed to characterize the signal generation and propagation media in order to develop an acoustic/seismic signal propagation model. Specifically, this paper discusses the coupling of acoustic energy in the soil as a function of frequency; the observation of signal amplitude variations as a function of distance and frequency; and the measurement of signal distortion through the calculation of the normalized correlation and coherence function. It has been found that acoustic/seismic cross properties provide good discrimination parameters and that seismic and acoustic differential propagation velocity and seismic multimode propagation velocity can be used to provide ranging information; that the signal amplitude is a direct function of the soil coupling properties and soil inhomogeneities; and that the acoustic/seismic, as well as the seismic propagation channel, is nonlinear and can be best characterized through the application of Weiner's nonlinear theory.
Acoustically induced instabilities of control valves
Authors: Ingard Uno
Published: October 2005

Recent theoretical and experimental studies of acoustically induced axial oscillations of control valves are reviewed, and a mechanism for acoustically induced lateral oscillations is proposed. Possible methods of quenching the instabilities are also discussed. [Work supported by ONR.]
Edge mode radiation due to a fluid-loading discontinuity
Authors: Pierucci M.
Published: October 2005

Edge mode radiation is determined for an infinite membrane with a discontinuous fluid loading condition. The membrane is assumed to be line driven. The lower portion of the membrane (y < 0) is exposed to a very light fluid which imposes no fluid leading on the structure. The upper half of the membrane (y > 0) is split up into two regimes (x < 0 and x > 0). The x < 0 regimes imposes fluid loading conditions on the membrane vibration while the x > 0 region is assumed to be a vacuum. The two upper regimes are separated by an acoustic anechoic surface. The radiated acoustic intensity as a function of frequency for different surface tensions and angular positions is derived and results are presented. [Portion of this work was sponsored by NAVSEA.]
Monaural detection of complex transient “real world” sounds
Authors: Price G. Richard , Hodge David C.
Published: October 2005

Given a long-term interest in evaluating auditory performance in the Army context, a first step was taken by examining the ability of individual ears to detect complex transient sounds, such as those produced by personnel movement or items of personnel combat equipment. A Fournier analysis was performed on each of 24 sounds every 20 msec and the energy was integrated into critical bands. In 20-msec steps within each critical band, the energy was integrated for 200- and the 200-msec period with the most energy in it was selected as the one most likely to be detected. These spectra were compared graphically with the auditory thresholds of 20 ears (selected to have a wide range of hearing levels) measured at critical-band center frequencies. The relative level at which the energy is one critical band equalled or exceeded the measured threshold for that critical band was selected and the detection levels were measured by having subjects track their thresholds for the complex sound. The correlations between the predicted and actual detection levels ranged from 0.89 to 0.98 with a 0.94 mean. Because of the generally broad distribution of energy in the transient sounds and the sensitivity of the ears in the midrange, most of the detections were a function of energy in this region. Ears with large high-frequency losses therefore did not perform much worse than the normal ears. If background noise typical of the world were included in the predictive procedure, then differences between ears would have been smaller.
Signal detection in time varying noise backgrounds
Authors: Fidell Sanford
Published: October 2005

Prediction of the detectability of well-defined signals in steady-state background noises are based on considerable theory and much practical experience. Little information is presently available to support predictions of detectability of complex signals in time-varying noise environments. The current study has temporal variability of background noise is long with respect to signal duration, observers' detection performance improves in more highly variable noise environments of constant average energy. Observers detecting a sinusoid embedded in one-third-octave band of Gaussian noise in a two alternative forced-choice task produced psychometric functions characterized by higher d′ values for increasingly variable background noises, relative to performance in steady-state noise.
Estimating the asymmetry of the auditory filter
Authors: Patterson Roy D. , Nimmo-Smith Ian
Published: October 2005

Although recent experiments have not revealed any marked asymmetry in the shape of the filter, some derivation assume a symmetric filter. To determine the degree of asymmetry more precisely, tone threshold was repeatedly measured as a notch in a wide-band noise was moved past the tone. The width of the notch was also varied. The asymmetry of the resulting threshold curves revealed a small asymmetry in the filter shape. In attempting to generalize earlier methods of estimating filter shape to the present data, it was discovered that the analytic statement of the problem is in a class mathematicians have labelled “ill-posed” because the “solution” filter is extremely sensitive to small perturbations of the threshold data. To improve the formulation of the problem a new and hopefully more useful class of filter shapes has been adopted.
Effect of stimulus variability on auditory filter shape
Authors: Patterson Roy D. , Henning G. Bruce
Published: October 2005

In recent derivations of auditory filter shape based on tone-in-noise detection experiments, the noise is represented by its power spectrum; that is, it has been assumed that filter shape is largely independent of masker variance. In an attempt to extend the generality of the filter-shape concept, one of the experiments has been replicated using a tonal maker in hopes of demonstrating that filter shapes derived with noise maskers are applicable when the masker variability is markedly reduced. Although some form of detection mechanism must follow the auditory filter, previous studies have not considered the mechanism explicitly. We will discuss the implications of including an energy detector at the output of the filter for the two-tone and noise-masking experiments, and in so doing, will outline the conditions under which changes in stimulus variability will affect performance and through it, the derived filter shapes.
Measurement of a binaural temporal transfer function
Authors: Grantham D. W. , Wightman F. L.
Published: October 2005

In order to derive a kind of “binaural temporal transfer function” for the auditory system, observers detected a stable target (a binaural click presented with an interaural time difference of +500 μsec) imbedded in a “moving” masker (a noise presented with an interaural delay sinusoidally varied from +500 to −500 μsec). Detectability of the click was measured as a function of the “position” of the noise at the time the click was presented and the rate at which the noise “moved”. As was expected, when the noise moved very slowly, less than 1 Hz, or not at all, detectability was best when the click appeared at one ear and the noise appeared at the other, and poorest when they both appeared at the same ear. At high rates (5 Hz) detectability was independent of the relative positions of the click and the noise. [This work was supported in part by NIH Grant NS12045.]
Structural rules for the perception of the periodicity of repeated random waveforms
Authors: Pollack Irwin
Published: October 2005

Two structural rules underlying the periodicity of repeated random waveforms were examined: the number of successive copies of the to-be-repeated elements; and phase inversion of the to-be-repeated elements. Two behavioral measures of periodicity strength were defined: the accuracy of pitch matching and the fraction of independently varying elements which must be inserted with a repeated random pattern for a threshold level of discrimination from a nonrandom pattern. A variety of autocorrelation measures were performed upon the signals. The signal measure most closely related to behavioral measures of periodicity strength is the maximum autocorrelation among the first 20 periodic delay intervals. Additional weightings with the average autocorrelation across the 20 delay intervals failed to improve further the relation with periodicity strength.
Detection of signals with identical energy spectra but different pitches under conditions of signal uncertainly
Authors: Lundeen C. , Small A. M.
Published: October 2005

Previous research has shown that a signal which is one of two pure tones will be less detectable than a signal that is of constant, known frequency throughout. Similar results were obtained in the present study using pulse train signals with identical energy spectra but different pitches. This finding cannot be attributed to the “mistuned” critical-band mechanism postulated in the single-band [W. P. Tanner Jr. and R. Z. Norman, IRE Trans. Prof. Group Inf. Theory PGIT-4, 222–227 (1954)] and multiple-band [D. M. Green, J. Acoust. Soc. Am. 30, 904–911 (1958)] models of selective auditory attention. Our subjects performed as if they could selectively attend to phase, and/or periodicity pitch cues in order to optimize the detectability of anticipated signals.
Dichotic perception of virtual pitch: Dependence on subjective lateral position of partials
Authors: Deutsch W. A. , Steger R.
Published: October 2005

Two general principles of pitch perception, analytic mode resulting in spectral pitch and synthetic mode resulting in virtual pitch have been investigated [E. J. Terhardt, Acoust. Soc. Am. 55, 1061–1069 (1974)]. It is assumed that virtual pitch is a product of Gestalt—perception, associated with some kind of learning process. If this concept is valid, it can be distinguished between spectral-pitch listeners (type I) and virtual-pitch listeners (type II). Complex tones consisting of components (a) 220, 330, 440, 550, and 660 Hz presented to the left ear and 550 and 660 Hz to the right ear were followed by (b) 200, 300, 400, 500, and 600 Hz left and 200, 400, and 600 right, which fused partially. The subjects were asked to decide whether pitch (a) or (b) was higher. Type I listeners responded (b) due to the spectral pitch of the corresponding lateral positions of the partials in the head, whereas type II listeners responded (a) referring to the missing fundamental of 110 Hz. In a second run the subjects were asked to pay specific attention to the different lateral positions too and again type I listeners differed from type II. The results are discussed in terms of Gestalt—rules and theory of pitch perception. [Work supported by AKG, Vienna.]
Left ear dichotic suppression following partial callosal section
Authors: Lenhardt M. L.
Published: October 2005

CV syllables (kindly provided by Kresge, LSU) were presented dichotically to a 13-year-old boy prior to and post surgical section of the majority of his corpus callosum. Pre-operatively no ear advantage was observed; however, after callosal section a clear right ear superiority emerged. Correct performance approximately doubled for the right ear and fell to change for the left ear. Voiceless CV's were reported correct more often than voiced, both prior to and post surgery. In addition, alveolars were more often reported correctly than labials, which in turn are reported more correct than velars. His single correct response performance was highest when pairs shared voicing preoperative and lowest when pairs shared place. Postoperatively sharing place still resulted in the lowest percent correct; however, performance was highest when pairs shared neither place nor voice. An analysis of pairs resulting in errors revealed that for the (1) voiceless/voiceless condition 16.6% errors occurred pre; and postoperatively, (2) voiced/voiced condition 21.0% errors occurred preop and 25.0% postop.; (3) voiceless/voiced condition 64.4% errors occurred preop and 58.4% postop. Thus error pattern did not change after section. These data suggest that callosal section results in suppression of the ear countralateral to the speech processor by removing the interhemisphere relay route.
Brainstem auditory-evoked responses with amplification
Authors: Mokotoff Barbara , Krebs Donald F.
Published: October 2005

Brainstem auditory-evoked response (BER) measurements were obtained on adult volunteer subjects who were hearing aid users. The BER's were elicited with click stimulation in several conditions: (1) unaided via earphones, (2) unaided via speaker, and (3) aided via speaker. In the aided condition, subjects adjusted their own hearing aids to their usual comfortable listening level. Aided and unaided audiometric thresholds were also obtained on each subject using pure tones, narrow-band noise, and speech under phones, and using noise and speech in a sound field. Hearing aids were analyzed for frequency response and distortion with a series of inputs and also at user level. Comparisons were made between aided and unaided audiological and BER data. Results indicate that aided BER measurement compared favorably with aided audiological data on cooperative adults and may therefore become useful when applied to hearing aid evaluation procedures for infants and children.
Seasonal dependence of ambient sea noise near the marginal ice zone of the Greenland sea
Authors: McGrath James R.
Published: October 2005

Omnidirectional ambient sea-noise measurements in the 20–60-Hz band near the marginal ice zone were recorded in the deepwater basin of the Greenland sea. This experiment, conducted between August 1972 and July 1973, provided 1-min samples of sea noise every 4 h using an omnidirectional hydrophone 306 m below the surface. During this time, the ice edge moved over the recording site, providing evidence that the iceline acts as a noise source, raising local levels above both open-ocean and ice-field values. Seasonal conditions favor the lowest ambient noise levels during midwinter (January) and early summer (June), while the highest levels occur during early spring (March) and early fall (November). Evidence indicates that the maximum ambient noise levels near the marginal ice zone are 12–16 dB higher than the maximum Arctic Ocean values under contiguous ice cover.
Effect of traffic mix volume and speed on highway noise levels
Authors: Peppin Richard J. , Manns Basil H.
Published: October 2005

Since Johnson and Saunders [“The Evaluation of Noise from Freely Flowing Traffic,” J. Sound Vib. 7, 287–309 (1968)] derived an expression relating the median traffic noise level as a function of traffic flow characteristics, many other relationships have emerged. These relationships have been used to design nomographs and computer noise prediction programs. This paper discusses the various basic formulation equations used to relate traffic mix and speed to the noise levels from motor vehicles. Parametric analysis of the basic equations used in the Revised Design Guide (NCHRP Report 3-7/3) for automobiles, medium trucks and heavy trucks, is performed. The effects of variations of traffic mix volume and speed on “equivalent” noise level at receivers are illustrated. Figures are plotted so that the results can be used to find the minimum noise levels for a given speed, based on a given traffic mix.
New applications of compliant tube technology
Authors: Brigham G.
Published: October 2005

The use of compliant tubes for pressure releasing in underwater electroacoustic transducers and for mechanical beamforming such as in acoustic lenses has been well known and publicized in the past 23 years. Less well known and more recent applications of tubing is in acoustic barriers/baffling, geodesic lensing/wave guiding, sonar array shading, sonar window trussing, and acoustic energy mode conversion such as in anechoic surfacing and/or wave guiding. While the basic acoustics/elastodynamics of tube arrays has been extensively developed to a point of near standardization in recent years, the use of the technology in any specific system still requires considerable numerical analysis. This paper discusses the newer applications and presents several examples of tube systems together with comparisons between analysis and measurements.
Automatic comparison calibration of hydrophones using the ambient noise background
Authors: Merklinger H. M.
Published: October 2005

A method is described which precisely and automatically compares the sensitivities of two hydrophones. The method is based on the property that the difference of two signals having equal average power is uncorrelated with the sum of the two signals. For two hydrophones placed in the same noise environment, a feedback circuit may be divised which adjusts the gain in one of the signal paths so that the required null correlation is achieved and maintained. The circuit may be analog, digital, or a hybrid combination of analog and digital. Frequency selectivity may be achieved with two identical filters. The method is particularly applicable to superdirective arrays since precise relative element weights may be achieved despite unknown or slowly changing hydrophone sensitivities. Under some circumstances data obtained with poorly weighted superdirective arrays may be improved after the telemetry and data recording operations have been completed.
Method of improving the performance of a multimode spherical hydrophone operating near a resonant frequency
Authors: Pond H. L. , Alatalo F. A. , Ko S. H.
Published: October 2005

A spherical elastic shell surrounded by fluid will be deformed by an imposed acoustic wave. If the shell is made of a piezoelectric material, a voltage is induced by the strains. Elements of shell insulated from one another act as hydophones. The responses of such a shell can be used to determine the direction of an incoming wave. A simple cosine pattern for normal deflection dominates from low frequencies through the resonant frequency for the first mode except for frequency bands near resonance for the third, fifth, and seventh modes. The simple detection of direction is based on a cosine pattern for the induced voltage. It is shown that the surface of the shell can be divided into selected areas which are insulated from one another and such that the sum of voltages from the elements of area is zero for a selected mode. This method is used to remove the effect of the third and most harmful mode and restore the simple cosine voltage pattern in the frequency band near the resonant frequency for the third mode. [Work supported by NAVMAT.]
Syllable timing as a function of position-in-utterance in infant babbling
Authors: Oller D. Kimbrough , Smith Bruce
Published: October 2005

For several years now it has been known that a number of languages show regular variations of syllable duration as a function of position-in-utterance. The most striking finding has been that final syllable vowels (in English, for instance) are up to 125 msec longer than comparable nonfinal syllable vowels. The present study attempts to determine whether or not the same sorts of patterns of syllable duration obtain in infant utterances. A corpus of reduplicated CVCV—utterances of several infants were tape-recorded and analyzed spectrographically. The durations of the infant consonants and vowels were compared with the durations of adult consonants and vowels as pronounced by adult speakers who were asked to read sequences phonetically like the infant utterances. The results show only a slight tendency toward final syllable lengthening in the infant utterances, a tendency which is demonstrably less than that which occurred in the comparable adult utterances. [Work supported by NIH-NICHD contract HD-3-2793 and Grant 5R01 HD 09906-02.]
Syntactic boundaries in the timing of trochaic speech
Authors: Cooper William E. , Lapointe Steven G. , Paccia Jeanne M.
Published: October 2005

The duration of a stressed syllable is shortened when it is immediately followed by an unstressed syllable. Previous work showed that this effect operates across word boundaries but is diminished in magnitude by the presence of an intervening syntactic boundary. In this study, the durations of key segments within stressed syllables were measured in sentence pairs containing a matched phonetic environment. The results for ten speakers showed that the shortening effect was blocked in the presence of a number of major syntactic boundaries, including the boundaries between a noun phrase (NP) and a prepositional phrase (PP), between two PP's, and between a NP or PP and a separate clause. Lesser syntactic boundaries, including the boundaries between a verb and NP direct object and between an NP direct object and NP indirect object, did not block the shortening rule. The magnitude of the blocking effect did not depend on the transformational history or internal structure of the syntactic constituents as much as on the boundary type.
Hearing “words” without words: speech prosody and word perception
Authors: Nakatani Lloyd H. , Schaffer Judith A.
Published: October 2005

The role of prosody in word perception was studied by asking listeners to locate word boundaries in speech with natural prosody but without phonotactic or semantic cues. To obtain such speech, seven speakers read sentences (e.g., “The noisy dog never stopped barking”) with “ma” substituted for each syllable of the three-syllable adjective-noun phrase (e.g., “The mama ma never stopped barking”). These nonsense phrases with all possible stress patterns were excised and played to 38 naive listeners who judged whether they heard each phrase as “ma mama” or “mama ma.” The stress pattern determined how well the phrases were parsed into words. When the stress pattern made the word division unambiguous (01-1, 1-12, 1-10 where 0, 2, 1 denote no stress, midstress, and high stress), listeners used the pattern to parse the phrases easily; striking individual differences were observed. When the stress pattern left the word division ambiguous (11-1, 1-11, 12-1, 1-21, 10-1, 1-01), parsing was harder but still better than chance. So not only global patterns, but also detailed information about prosody, are used in word perception.
How is the memory search of a two-clause complex sentence in immediate memory carried out?
Authors: Shedletsky Leonard
Published: October 2005

An item recognition task was performed with 32 native English speaking, adult, right-handed subjects who listened to eight two-clause complex sentences presented to the left ear, each immediately followed by a probe word presented to the right ear. The subjects indicated whether or not the probe word occurred in the sentence and their recognition latency was measured. An analysis of variance was performed on recognition latency as a function of the three independent variables: (a) the serial position of the target word, early or late, within (b) a main or subordinate clause, in (c) initial or final clause position. The findings of this experiment were: (a) a word in the final clause is recognized significantly faster than a word in the initial clause; (b) for subordinate clauses, subjects take longer to respond to a target word occurring late in the clause than to a target word occurring early in the clause; for main clauses, subjects take longer to respond to a target word occurring early in the clause than to a target word occurring late in the clause. Present storage models of sentence processing and memory search models are inadequate to account for all the data. A combined storage-search account was proposed. A serial self-terminating model of clause accessing, with final clause search occurring prior to initial clause search, fit the data better than a simultaneous search of both clauses. Clauses are searched either in a primary or a secondary buffer, depending on clause type (main or subordinate) and clause position (initial or final) in the sentence. To explain the difference in mode of search between main and subordinate clauses, it was suggested that main clauses exhibit a property of primacy over subordinate clauses.
Acoustic echo sounding of the marine boundary layer
Authors: Noonkester V. R. , Jensen D. R. , Richter J. H.
Published: October 2005

Acoustical probing of the lower atmosphere has proliferated since 1968 when McAllister [J. Atmos. Terr. Phys. 30, 1439–1440 (1968)] demonstrated its advantages. Considerable acoustic echo soundings have been made in continental regions but echo sounding of the marine boundary layer (MBL) is fragmentary. Many hours of acoustic echo-sounding data have been taken since 1974 at a multisensor site on the Pacific coast in San Diego, CA, where the prevailing northwesterly winds carry the MBL over the echo sounder almost continuously. Simultaneous observations of the MBL by the echo sounder, other surface-based remote sensors (e.g., FM-CW radar and lidar) and standard meterological sensors (e.g., radiosondes) have aided in the interpretation of the echo-sounder data. These observations have been augmented by observations on the Atlantic coast near Wallops Island, VA. The paper will present some characteristic echo-sounder data of the MBL with meterological interpretations.
Acoustic sounding of the tropical marine boundary layer during GATE
Authors: Mandics P. A. , Gaynor J. E. , Hall F. F.
Published: October 2005

A vertically pointed monostatic acoustic sounder was installed on the NOAA Ship OCEANOGRAPHER during the Global Atmospheric Research Program Atlantic Tropical Experiment (GATE). The sounder antenna was mounted on a gyrocontrolled platform to compensate for the ship's pitch and roll motions. Extensive measures such as mounting the antenna assembly on a vibration isolator and installing absorbing cuffs had to be taken to reduce interference by ship-generated noise. Back-scattered acoustic data obtained from up to 850 m height describe the tropical marine boundary layer in unique and hitherto unseen detail. During undisturbed weather conditions the facsimile record showed convective plumes rising from the surface of the water up to 400 m. Storm-generated disturbances resulted in a substantial modification of the boundary layer; low-level, multilayered, undulating inversions formed from cool outflow currents. The inversions persisted for up to 16 hours. Low-level, patchy cumulus clouds produced characteristic hummock-shaped acoustic echoes. Analysis of the Doppler frequency shift of the returns allowed the determination of vertical velocities within these clouds and underlying convective plumes.