bluuebunny/crossref_metadata_2025 huggingface dataset:
The Rotorcraft Noise Model (RNM)
Authors: Conner David , Burley Casey , Smith Charles , Pope Dennis
Published: September 2008

The Rotorcraft Noise Model (RNM) is a simulation program that predicts how sound will propagate through the atmosphere and accumulate at receiver locations located on flat ground or varying terrain, for single and multiple vehicle flight operations. Developed by the NASA Langley Research Center and Wyle Laboratories, RNM was initially intended to provide a tool to aid in the development of low noise terminal area operations for rotorcraft and tiltrotors, but has since been widely adopted for environmental impact studies. At the core of RNM are the vehicle source noise characteristics, input as sound hemispheres. As the vehicle flies along its prescribed flight trajectory, the source noise propagation is simulated and accumulated at the receiver locations (single points of interest or multiple grid points) in a systematic time-based manner. These sound signals at the receiver locations may then be analyzed to obtain numerous ground metrics or noise animation videos. A PC executable version of RNM is publicly available. This presentation will provide details on the theoretical framework of RNMs propagation model and the required inputs, discussion on capabilities for civil and military applications as well as details on output results. Code validation results and example outputs will be presented.
Advances in distributed arrays for detection of infrasonic events
Authors: Dillion Kevin , Howard Wheeler , Shields F. Douglas
Published: September 2008

Infrasound arrays normally consist of four to eight microbarometers spaced kilometers apart. Each of these microbarometers is connected to a pipe or porous hose array to reduce wind noise. This presentation describes a 100 sensor all weather distributed array and its use in continuously logging infrasound signals over an extended period of time. The array has been deployed next to a conventional porous hose array connected to a Chaparral 2.5 microbarometer. The effectiveness of the two arrays in canceling wind noise and detecting infrasound signals is compared. The hardware and software used by the distributed array to log the data and to process it so as to cancel wind noise, identify and separate the infrasound signals, and locate their sources will be discussed.
Computing pitch of speech and music using a sawtooth waveform inspired pitch estimator
Authors: Camacho Arturo , Harris John G.
Published: September 2008

A powerful pitch estimation algorithm called SWIPE has been developed for processing speech and music. SWIPE is shown to outperform existing algorithms on several publicly available speech and musical instrument databases, and a disordered speech database, reducing the gross error rate by 40%, relative to the best competing algorithm. In short, SWIPE estimates the pitch as the fundamental frequency of a sawtooth waveform, whose spectrum best matches the spectrum of the input signal. The short-time Fourier transform of the sawtooth waveform provides an extension to older frequency-based, sieve-type estimation algorithms by providing smooth peaks with decaying amplitudes to correlate with the fundamental frequency (if present) and its harmonics. An improvement on the algorithm is achieved by using only the first and prime harmonics, which significantly reduces subharmonic errors commonly found in other pitch estimation algorithms.
Error correction and accuracy scaling for a joint subtraction multiple fundamental frequency estimation algorithm
Authors: Herman Geoffrey , Beauchamp James
Published: September 2008

This talk presents an improved multiple fundamental frequency (F0) estimation algorithm, which is loosely based on a previously developed joint subtraction algorithm [A. Klapuri, Proc. 7th Int. Conf. on Music Information Retrieval, 216–221 (2006)]. The improved algorithm uses some auditory scene analysis techniques to supplement a maximum likelihood based F0 estimator to predict the best combination of F0s. The accuracy of this base algorithm is then improved by use of error detection/correction. Error rates of the algorithm were collected using random mixes of sounds gathered from the University of Iowa Theremin database of musical instrument samples, as well as mixes of continuous audio. The performance of both the base algorithm and the improved algorithm will be presented together. It was found that the improved algorithm decreases error rates by as much as 50%, compared to the base algorithm with minimal loss in computational efficiency. The trade-offs between computational efficiency and accuracy, tested via a tunable parameter built into the algorithm, will be discussed. Trade-offs between computational costs and accuracy as a function of the number of F0s will also be discussed.
A method for multiple F0 detection of continuous audio with dissimilar timbres
Authors: Bay Mert , Beauchamp James W.
Published: September 2008

This paper discusses a method for multiple fundamental frequency (F0) tracking performed on a polyphonic mixture of musical instruments with dissimilar timbres recorded on a single monaural track. For each time frame, given a number of possible F0 candidates, F0 trajectories are extracted using an HMM-like method where observations are the input spectra and each F0 candidate is represented by one of the hidden states. The F0 trajectories are optimized according to a harmonic overlap score for each F0 candidate and a continuity score that is based on a distance between the F0 candidate’s harmonic amplitudes and frequencies calculated for all possible candidate pairs at successive adjacent frames. This step leads to the formation of note tracks. In a second step, instrument identification is performed on harmonic spectra associated with these note tracks to group similar timbres together. The method is demonstrated with an acoustic audio recording of a woodwind quintet. Applications include audio-to-MIDI conversion, instrument/voice separation, and automatic music transcription. F0 tracks can also be used as high level features for music information retrieval systems.
Towards estimation of the number of concurrent F0s
Authors: Yeh Chunghsin , Roebel Axel
Published: September 2008

Two central problems common to many multiple F0 estimation algorithms will be presented: The estimation and handling of the noise level and the modeling and representation of overlapped partials. Contrary to the widely used assumption that the noise is white Gaussian, it is proposed to use an adaptive and frequency-dependent noise level to be able to improve the description of sinusoidal partials and the expected residual. By using the frequency-dependent noise level, an F0 estimation algorithm can be guided to extract sources having designated harmonic-to-noise ratios. One of the major problems in multiple F0 estimation is partial collision. The assumption that is usually made to estimate the expected amplitudes of overlapped partials is the additivity of magnitude spectra or the additivity of power spectra. A mathematical investigation into the properties of overlapped amplitudes allows us to present an improved probabilistic description of the expected amplitudes. The implications of these two refinements for the estimation of the number of sources that are present in a recording will be discussed, and preliminary experimental results will be presented.
A new strategy on pitch estimation in polyphonic musical audio
Authors: Dressler Karin
Published: September 2008

In this paper, a new perceptually motivated approach for pitch estimation in polyphonic musical audio is presented. The proposed algorithm is based on the pair-wise analysis of spectral peaks. Starting from a spectral representation of the audio signal, audible spectral peaks are selected with the help of a simplified psychoacoustic model. The idea of the technique lies in the identification of partials with a successive harmonic number (1 and 2, 2 and 3, etc.). Since two successive harmonics of a harmonic sound have a well-defined frequency ratio, a possible fundamental can be derived from the instantaneous frequencies of the two peaks. Furthermore, under certain conditions the harmonic integer ratios 5/3, 7/5, and 9/7 are also considered. Consecutively, the identified harmonic pairs are evaluated according to a perceptually motivated rating scheme, and the resulting pitch strengths are added to a pitch spectrogram. In addition to the efficient estimation of the fundamental frequency, the presented approach results in a quantitative prediction of the perceived pitch strength, allowing the identification of the predominant voices (e.g., melody and bass line) in music recordings. The method is robust in regard to the missing fundamental frequency phenomenon and works well with quasi-harmonic sounds.
An approach for melody extraction from polyphonic audio: Using perceptual principles and melodic smoothness
Authors: Paiva Rui P.
Published: September 2008

In this research work, the problem of melody extraction from polyphonic audio is addressed. A multi-stage approach is followed, inspired on principles from perceptual theory and musical practice. Physiological models and perceptual cues of sound organization are incorporated into the method, mimicing the behavior of the human auditory system to some extent. Moreover, musicological principles are applied, in order to support the identification of the musical notes that convey the main melodic line. The system comprises three main modules, where a number of rule-based procedures are proposed: (i) Pitch detection, where an auditory model-based pitch detector is employed for selecting multiple pitches in each analysis frame; (ii) determination of musical notes (with precise temporal boundaries and pitches); and (iii) identification of melodic notes, based on two core assumptions that we designate as the salience principle and the melodic smoothness principle. Experimental results were conducted, showing that the method performs satisfactorily under the specified assumptions, namely when the notes comprising the melody are, in general, more intense than the accompanying instruments. However, additional difficulties are encountered in song excerpts where the intensity of the melody in comparison to the surrounding accompaniment is not so favorable.
Aero-acoustic noise generation of a lawnmower deck
Authors: Peak Chris
Published: September 2008

Sound power and dynamic pressure measurements were used to identify where noise is generated by a single spindle lawnmower deck. The sound power measurements were performed with different blades and deck profiles to quantify the overall levels. An electric motor was used to turn the blade, and the measurements were made in a semi-anechoic chamber over an absorptive surface. The test procedures outlined in ISO 11094 were used as a guideline. The microphones used for the dynamic pressure measurements were located on a two-inch by two-inch grid, one inch below the cutting plane of the mower. The dynamic pressure data were used to generate one-third octave colormaps and movies of the pressure distribution under the mower deck. The colormaps and movies identified local hot spots where noise is generated, and they were able to distinguish between blade and deck configurations with a difference of two dB in the sound power level. [The work was completed to fulfill the requirements for a Masters Degree from The Pennsylvania State University Graduate Program in Acoustics. Dr. Jerry Lauchle was my advisor for the project.]
Computational aeroacoustic modeling of rotary mower noise
Authors: Washburn Karl , Peak Christopher , Sandboge Robert
Published: September 2008

The primary source of noise from rotary mowing machines is the aeroacoustic sound generated by the cutting blades. The requirements for maintaining high quality of the cut contradict those for reducing noise from the blade. The key to balancing these competing requirements is understanding the aeroacoustics of blades within the cutting environment. Until recently, there has been no reliable way to predict the noise generated by a configuration of blades and housing decks. To address this, John Deere and its partners have undertaken a program in Computational Aero-Acoustics (CAA). By leveraging existing computational fluid dynamics and parallel processing capabilities, the participation of several software vendors, and the cooperation of industrial and academic partners, this project has provided a capability to accurately predict the sound generated by low Mach number flows in industrial systems. This presentation reviews the results of applying a nascent CAA capability. The noise generated by a single-blade walk-behind mower was computed and compares successfully with laboratory tests of radiated sound from the mower. Details of the computational process and the results will be presented.
Comparison of the conical beam and image source methods for predicting sound fields in a branching tunnel
Authors: So Sai Tung , Li Kai Ming
Published: September 2008

There are many different models for the numerical analysis of a tunnel with a branch, i.e., a T-intersection. Typical numerical models include the finite-element method, boundary-element method, conical beam, ray-tracing approach, and the image-source method. Among these different models, the image-source method is one of the simplest and requires the least computational time. In this study, the prediction of sound transmission along a branching underground tunnel is computed by using the image-source method. There were published data for the measured sound fields along a branching underground tunnel with an arched ceiling. Numerical predictions of sound transmission by means of the conical-beam method were also presented in this earlier study. By applying the information extracted from the published data, the image-source method was applied to predict the transmission of sound for an identical arrangement, except that the tunnel was assumed to have a rectangular cross section. The reverberation time and rapid speech transmission index of this branching tunnel were also computed by means of the image-source method. It was demonstrated that the predicted results according to the image source method compared reasonably well with the published experimental results.
In-duct acoustic source model considering flow characteristics at orifice and valve inside fluid machinery
Authors: Kim Tae-Kyoon , Ih Jeong-Guon
Published: September 2008

Intake and exhaust systems of fluid machinery are generally nonlinear and time-varying due to the flow activities at orifice, periodically moving valve, etc. Because of difficulties in considering these effects in the analysis of the sound propagation, a linear time-invariant source model has been used. Negative source resistance and spectral errors at high frequencies of a linear time-invariant model are due to the neglect of such reality. To investigate the effect of nonlinear and time-varying nature of the in-duct source, flow characteristics through orifice and periodic change of valve area were modeled. A simplified fluid machinery model composed of reservoir, valve, and duct was tested. To obtain a reference acoustic characteristic at the source plane, the method of characteristics (MOC) was employed. Pressure spectrum at the source plane using the present model was compared with those from a linear time-invariant source model and MOC. At the fundamental frequency and low harmonics of valve frequency, the pressure spectrum has little difference in both models, which results in a similar overall sound level. However, at high harmonics, the suggested model showed large discrepancy with the linear time-invariant model, but was similar to the MOC result. This revealed a good possibility of the suggested model for the practical application to the fluid machinery with further extended study. [Work supported by BK21.]
Simpified theory of wind turbine noise
Authors: Kotsari Evangelia G. , Pierce Allan D.
Published: September 2008

The noise is expected to be accounted for by the forces exerted on the air by the blades and pillar; the idealized design is presumed sufficiently optimal that the power is maximum and the generated sound is minimal for that power generation. Betz’s law applies, and the steady torque exerted on the air by the turbine has a relation to the net drag force (in the direction of incoming flow) exerted on the turbine blade system. The forces associated with the torque create an acoustic pressure field that is even in the coordinate along the turbine axis. The forces associated with the drag create a field that is odd in this coordinate. These two contributions can constructively reinforce or interfere, so the various pure tone contributions at multiples of the blade passage frequency can have nodes in the polar plot of radiated far-field intensity. The locations of these are explained with the use of Betz’s law. At moderate to higher frequencies, interest is typically in finite-frequency bands, which contain a large number of discrete tones. To predict the noise within such bands, sums that appear in the basic Gutin theory are replaced by integrals, and appropriate smoothing techniques are applied.
Environmental noise measurements using a coded signal
Authors: Waddington David C. , Angus Jamie A. , Lam Yui W.
Published: September 2008

Application of a coded acoustic signal to the investigation of environmental sound propagation is presented. The problem identified is that of obtaining adequate signal-to-noise ratio in environmental measurements, since the sound power level of the source is often required to be limited so as not to cause a noise disturbance. An established method of increasing signal-to-noise ratio is by means of coded signals such as maximum length sequences. However, this method cannot be applied outdoors because of the time-variant nature of the atmosphere. Consequently, a coded acoustic signal design is proposed for the study of environmental sound propagation, involving the modulation of multiple acoustic carriers by orthogonal correlation sequences. An experiment is described in which the correlation properties of the coded signal are demonstrated under outdoor conditions typical of environmental noise measurements. The results illustrate that a coded signal approach can provide researchers, engineers, and environmental officers with an improved method for investigating the propagation of environmental noise. This raises the possibility that investigations could be performed in the urban environment without disturbance to residents, and propagation could be determined in the presence of high background noise, such as from roads. [Work supported by EPSRC UK.]
Measurement and evaluation of blade passage frequency fluctuations
Authors: Duke Cole V. , Sommerfeldt Scott D. , Gee Kent L. , Duke Connor R.
Published: September 2008

In the active control of tonal noise from cooling fans, one factor that can limit the achievable attenuation is fluctuation of the blade passage frequency in time. Large fluctuations in a short time can hinder the algorithm from converging to the optimal solution. Some fans have steadier speeds than others, which can be due to unsteady driving mechanisms or the physical structure of the fan. Environmental effects, such as back pressure and unsteady blade loading, can also cause the fan speed to fluctuate. The shifting in the blade passage frequency will be measured using a zero-crossing technique to track the frequency of each cycle. Blade passage frequency fluctuations will be presented for various driving mechanisms and environmental conditions. Techniques to minimize frequency shifting will also be discussed.
A fast method of deriving the Kirchhoff formula for moving surfaces
Authors: Farassat F. , Posey J.
Published: September 2008

The Kirchhoff formula for a moving surface is very useful in many wave propagation problems, particularly in the prediction of noise from rotating machinery. Several publications in the last two decades have presented derivations of the Kirchhoff formula for moving surfaces in both time and frequency domains. The method presented here, originally developed by Farassat and Myers in time domain, is both simple and direct. It is based on generalized function theory and the useful concept of imbedding the problem in the unbounded three-dimensional space. An inhomogeneous wave equation is derived with source terms that involve Dirac delta functions with their supports on the moving data surface. This wave equation is then solved using the simple free space Green’s function of the wave equation resulting in the Kirchhoff formula. The algebraic manipulations are minimal and simple. The derivation does not require the Green’s theorem in four dimensions and there is no ambiguity in the interpretation of any terms in the final formulas. Furthermore, this method also gives the simplest derivation of the classical Kirchhoff formula which has a fairly lengthy derivation in physics and applied mathematics books. The Farassat-Myers method can be used easily in frequency domain.
The effects of frequency and sound pressure level on acoustic agglomeration of flyash
Authors: Lipkens Bart , Dong Shaozeng
Published: September 2008

Previously, a new concept of effective agglomeration length, which measures the maximum particle separation distance for effective collisions, was proposed (J. Aerosol Sc., Vol. 37, 540–553, 2006) for acoustic agglomeration of polydisperse aerosols. Two mechanisms are taken into account, orthokinetic collision and acoustic wake in a horizontal acoustic wave. Predictions of the model indicate that orthokinetic collision dominates at low frequencies for intermediate particle size ratios while the acoustic wake effect is more significant at higher frequencies for all particle sizes. Experimental results of acoustic agglomeration of flyash in two resonators, one a constant cross-sectional resonator and the second a variable cross-sectional resonator, are reported. The effects of frequency and sound pressure level (SPL) on acoustic agglomeration are measured. The experimental results are compared with predictions from the effective agglomeration length model. It is confirmed that for orthokinetic agglomeration an optimum frequency exists. However, the optimum frequency is found to be a function of SPL. Resonator design has an effect on the maximum achievable SPL and the power consumption of the device.
Computer simulations of a maximum length sequence modulated photoacoustic spectrometer
Authors: Soleimani-Karimabad Arash , Muehleisen Ralph T.
Published: September 2008

There is a great need for inexpensive, rugged, portable, and versatile chemical detectors for use in both security and environmental measurement, but current technologies that meet that criterion are usually limited in sensitivity and/or are dedicated to detecting a limited number of chemicals. Photoacoustic spectrometers are among the most sensitive instruments for trace analysis in gases, liquids, and on surfaces, and many are able to detect a wide range of different substances. Unfortunately, photoacoustic systems are usually expensive, are usually not portable, and usually cannot be used for both surface and airborne measurements. A new photoacoustic measurement system has been proposed that utilizes maximum length sequence (MLS) modulation of small, rugged, inexpensive laser diodes. The use of MLS modulation eliminates the need for lock-in amplifiers, optical modulators, or resonant measurement cells allowing for portable use on both gas and surfaces. In this paper, results of computer simulations to examine the sensitivity limits and noise immunity of the proposed design are presented.
Anisotropic dynamical mass density by two-dimensional arrays of solid cylinders in air
Authors: Torrent Daniel , Sánchez-Dehesa José
Published: September 2008

The long wavelength limit (homogenization) of two-dimensional (2-D) periodic arrangements of elastic cylinders in a fluid or a gas (i.e., sonic crystals) is analyzed through the linear part of their band structure. It is found that sonic crystals behave (in the homogenization limit) as effective metamaterials with anisotropic dynamical mass density and fluidlike speed of sound. The special case of rigid cylinders embedded in several 2-D lattices is studied here. It is shown that these systems are suitable to build 2-D anisotropic fluidlike metamaterials. Finally, the reflection and transmission through an anisotropic-elastic medium will be discussed. [Work supported by MEC of Spain and GVA of Valencia.]
Probing the temporal resolution and bandwidth of human hearing
Authors: Kunchur Milind N.
Published: September 2008

Two types of experiments were conducted to assess the human discriminability of temporal convolution—one employed lowpass filtering and the other employed delays due to spatial misalignment. By using special ultrahigh-fidelity equipment, both experiments demonstrated discernment at a ∼5μs timescale, which is much shorter than found previously. While the signal manipulations affect both the spectrum and temporal definition of the signal, the spectral changes in these experiments fall below the known just noticeable differences. The discrimination may therefore involve mechanisms additional to the auditory system’s ability to distinguish spectral amplitude differences. Furthemore the present work shows that typical instrumentation used in psychoacoustic research has insufficient temporal speed and fidelity for properly assessing all aspects of human hearing and also that digital sampling rates used in consumer audio are insufficient for fully preserving transparency. [This work was partially supported by the University of South Carolina Office of Research and Health Sciences Research Funding Program.]
Unmasking the components of a complex tone with stretched harmonics
Authors: Hartmann William M.
Published: September 2008

In the unmasking paradigm [W.M. Hartmann and M.J. Goupell, J. Acoust. Soc. Am. 120, 2142–2157 (2006)], removing harmonic n of a periodic complex tone causes harmonic n+1 to be suddenly audible, though sometimes n−1 is heard instead. The unmasking experiment was repeated with harmonically stretched tones, where the frequency of the nth ‘‘harmonic’’ was proportional to the 1.32 power of n (octave ratio of 2.5 to 1). After a six-interval sequence with component n alternately on and off (1<n<21), listeners matched the pitch of the unmasked component by adjusting the frequency of a sine tone. Pitch matching results were comparable to results for harmonic tones, showing regions of: (1) Accurate matches; (2) reproducible pitch shifts; and (3) inconsistent matches as the removed component frequency increased. The agreement in results proves that unmasking does not depend on harmonicity. It also indicates that pitch matches were not biased by the listener’s internal template for the harmonic series in any important way. Because of the stretch in component frequencies, listeners were able to make reliable matches at higher frequencies, as high as 3,500 Hz, demonstrating the importance of frequency resolution in the unmasking effect. [Work supported by the NIDCD Grant No. DC00181.]
Context effects in the lateralization of sine tones?
Authors: Hartmann William M. , Nolan Erin M.
Published: September 2008

Low-frequency sine tones can have large interaural time differences, but high-frequency tones cannot. If listeners estimate the lateral positions of tones of different frequency, using a fixed lateralization response scale, the estimates may depend on the experimental blocking of the tones. A context where all frequencies appear equally often (mixed) serves as a standard. It was hypothesized that a context with tones blocked on frequency would lead to abnormally large lateral position estimates for high-frequency tones because of a tendency to use the entire response scale in any stimulus-range context. Four listeners estimated the lateral positions of sine tones with frequencies 200, 500, 750, and 1,000 Hz having all possible interaural phase differences (IPD) spaced by 30 deg. Fixed-frequency blocks were done first, then mixed. Very little trace of the hypothesized effect was observed, either in the overall range of responses or in the slope of the response-IPD function for small IPD values. The same listeners matched the laterality of the same tones using the interaural level difference (ILD) of a narrow noise band. Relative slopes of the ILD-IPD functions agreed with slopes of the lateral position estimates. [Work supported by the NIDCD.]
Anomalyous interaural coherence minima near 45 degrees
Authors: Aaronson Neil L. , Wolf Eric
Published: September 2008

High-resolution (200 kHz sample rate) measurements of interaural coherence for a KEMAR manikin were made in an anechoic environment. As the angle of incident sound was varied in the horizontal plane, there was a minimum in coherence around ±45°. These coherence minima occured at low frequencies, up to about 3000 Hz, and decreased in depth for increasing frequency, f, at a rate approximately proportional to 1/f. In a 1/3-octave band centered at 500 Hz, the interaural coherence was approximately 0.7 for angles of ±45°. These results were confirmed by measurements in KEMAR using probe microphones. Measurements made using probe microphones with human listeners at angles of ±45° and 0° showed a similar pattern in coherences, but the minima were far more shallow. In the 500 Hz band, a coherence of, on average, 0.94 was measured in listeners facing ±45°, compared to an average coherence of 0.99 in that band at 0°. Headphone experiments using measured values of coherence showed that the deeper measured coherence minima lead to a deficit in sound localization. [Work supported by the NIDCD grant DC00181.]
Sound scattering from two concentric fluid spheres
Authors: McNew Jared , O’Brien William D.
Published: September 2008

Recently, the finite-element method has been used to analyze the paths through which sound enters the human head. In this work, the solution to the problem of scattering by two concentric fluid spheres is derived in order to verify this model. The effect of differences in sound speed and density is taken into consideration while dissipation is neglected. The concentric sphere solution is then used as a simplified model of the human head where the outer sphere has the properties of bone and the inner sphere has the properties of brain. A Fourier cosine series is used to approximate a pulse propagating into the concentric sphere model. A correlation-based wavefront reconstruction is then performed, followed by ray tracing to visualize the results. [Work supported by AFOSR FA9550-06-0128.]
New procedure for measuring stimulus frequency otoacoustic emissions
Authors: Long Glenis R. , Talmadge Carrick L.
Published: September 2008

Stimulus frequency otoacoustic emissions (SFOAE) are generated by single tones. This is both an advantage and a disadvantage. The advantage is that they are more likely to be primarily generated from a more limited region of the cochlea than other types of OAE. The disadvantage is that when tone pulses are used, they occur at the same frequency and time as the stimulus, making it very difficult to separate the OAE from the signal, unless one uses suppression. The SFOAE is generated in the cochlea and has to travel along the basilar membrane before being measured in the ear canal. Cochlear travel times mean that the SFOAE is always delayed relative to the stimulus. The delay is frequency dependent. We have modified the continuously sweeping tone procedure developed in our lab for evaluating DPOAE to separate the stimulus and SFOAE. The LSF procedure was modified to permit dynamic fitting of the delay of a given signal component permitting the extraction of the time varying SFOAE.
Orthotropic material properties of the gerbil basilar membrane
Authors: Liu Shuangqin , White Robert D.
Published: September 2008

Material properties of the cochlear partition play an important role in understanding the cochlear hydrodynamics. In this paper, two sets of experimental results are analyzed to extract the orthotropic material properties of the gerbil cochlear partition at different longitudinal locations. In order to accomplish this, a geometrically nonlinear composite orthotropic plate model is employed. The model is used to predict both out-of-plane and in-plane motion of the partition under a static finite area distributed load. Both in-plane and out-of-plane motion are needed for comparison with recent experimental results. It is shown that the spatial decay rate (the space constant) for the in-plane deflection is different from the out-of-plane deflection, which has a significant effect on the derived partition properties. The size of the probe tip is shown to have little influence on the results. Results are presented for two types of boundary conditions. Orthotropy ratios (the ratio of the two elastic moduli) determined from the experimental data are found to vary with longitudinal position and choice of boundary conditions. Orthotropy ratios vary from 65 close to the base to 10 in the upper middle turn of the cochlea.
Mechanisms of auditory selective attention
Authors: Melara Robert
Published: September 2008

Recent research on auditory attention addresses three main questions: (1) How early in the processing stream does attention influence the analysis of auditory signals?; (2) are the distinct features of an auditory object processed together or separately?; and (3) how does the attentional system enable task-relevant information to be analyzed separately from task-irrelevant information? The three questions are answered here using behavioral and electrophysiological evidence. First, psychophysical evidence suggests that attention operates early, modulating auditory filters (critical bands). Yet, little physiological evidence has been found for attentional effects on a peripheral auditory system. Such evidence suggests instead that the effects of attention are confined mainly to cortical regions outside the primary auditory cortex. Second, information theory and signal-detection models suggest that auditory dimensions are integral, with features quickly bound into auditory objects. However, physiological evidence reveals that separate auditory features are processed in parallel and exhaustively from the earliest stages. Finally, electrophysiological evidence indicates that auditory selection involves a duality of processes: Excitation of task-relevant features and inhibition of task-irrelevant features. The magnitude of selection failure is mainly determined by the degree of psychophysical change in distractors (tectonic theory), rather than their physical similarity to targets (biased competition and attentional-trace theory).
Selective attention to time scales and time points during speech perception
Authors: Sanders Lisa D.
Published: September 2008

Selective attention provides a mechanism by which people preferentially process subsets of stimuli when faced with overwhelming amounts of information. Spatially selective attention is particularly important for perceiving complex scenes in which multiple objects are presented simultaneously at distinct locations. Recent evidence indicates that temporally selective attention plays an important role in perception when more information than can be processed in detail is presented rapidly. Specifically, listeners can direct selective attention to both different levels of hierarchically organized rhythms and to distinct time points. Doing so modulates auditory evoked potentials at early perceptual stages. Additional studies show that directing temporally selective attention to hierarchical levels and specific time points affects perception of the rapidly changing acoustic features that constitute speech. Language experience and proficiency affect the skill with which adult listeners select the most relevant features and differentially process attended and unattended information in speech streams. Furthermore, developmental deficits in selective attention have been implicated in specific language impairment. These deficits can be reduced with interventions designed to improve speech perception and are accompanied by increases in receptive language scores. [Work supported by NIDCD grants R03DC008684 and F32DC005291 to LDS, R01DC005660 to David Poeppel, and R01DC00128 and R01DC00481 to Helen Neville.]
Determinants of auditory cue weighting and selective attention
Authors: Holt Lori L. , Lotto Andrew J.
Published: September 2008

The categorization of speech sounds into phonetic units has been characterized as involving the differential weighting of perceptual cues. Cue weighting has been often explicitly or implicitly associated with the construct of selective attention. We will present data from speech and non-speech categorization studies that attempt to illuminate the factors that influence relative weighting of perceptual dimensions. Of particular interest is the role of variability across the different auditory dimensions. The results of these experiments will be related to work on cue weighting in other psychophysical tasks and the usefulness of the attention construct will be explored. [Work supported by NSF BCS0345773].
The behavior of affricates under restricted lengthening conditions
Authors: Pycha Anne
Published: September 2008

This study uses affricates as a test case for lengthening processes in speech. In general, consonants can lengthen as the result of phonological factors, such as a geminating suffix, or as the result of phonetic factors such as position within a phrase (Keating et al. 2003, Byrd et al. 2000, 2005). In Hungarian and many other languages, however, the presence of an additional consonant restricts phonological lengthening because CCC clusters containing geminates are ill-formed (Kenesei et al. 1998). Acoustic duration measurements were used to examine whether this restriction also holds for phonetic lengthening. Initial results from three male Hungarian speakers indicate that it does not. Consonants in phrase-final position are significantly longer than their counterparts in phrase-medial position, whether they stand alone or in a cluster. A focus on lengthened affricates, however, reveals that the ‘‘shape’’ of lengthening differs in these two conditions. Affricates in the stand-alone condition lengthen in a lopsided fashion, showing increases primarily in frication duration; those in the cluster condition lengthen more symmetrically, with increases in both stop closure and frication durations. These findings provide new support for the idea that affricates possess different structures at different levels of analysis (Lombardi 1990, Clements 1999).
Intelligibility of information in temporally desynchronized bands of speech
Authors: Stilp Christian E. , Kluender Keith R. , Alexander Joshua M. , Kiefte Michael
Published: September 2008

[Stilp et al., [J. Acoust. Soc. Am. 121, 3195 (2007)] demonstrated that acoustic measures of potential information, absent explicit linguistic information, are closely related to intelligibility of locally time-reversed sentences. Here, this finding is replicated in two experiments in which synchronization of speech energy within four spectral bands was perturbed in different ways. An ensemble of seven-syllable sentences was synthesized at three different speaking rates (2.5, 5.0, 10 syllables/s) and filtered by one-third-octave Butterworth filters centered at 335, 850, 2135, and 5400 Hz. In Experiment 1, onset of the 335- and 5,400-Hz bands was delayed by multiple durations (0, 25, 50, 100, 200, 400, 600 ms). Across speaking rates, intelligibility was a nonmonotonic function of delay with local minima corresponding to roughly one syllable. Smaller and larger delays yielded better performance. In Experiment 2, bands were nonlinearly scaled in time (jittered) to preserve overall duration with local rate adjusted to preserve the same overall degree of desynchronization as Experiment 1. Performance monotonically decreased with increasing jitter. Overall, results closely mimic previous findings for locally time-reserved sentences, and provide support that physical acoustic potential information closely relates to intelligibility of temporally distorted speech. [Work supported by NIDCD.]
Determining and interpreting acoustic landmark sequences in American English
Authors: Park Chiyoun , Chen Nancy , Jung Youngsook
Published: September 2008

Acoustically-significant time points (i.e., landmarks) can be used in knowledge-based speech recognition. Consonant landmarks represent three types of speech production processes (glottal vibration, sonorant, and obstruent consonant [K.N. Stevens, J. Acoust. Soc. Am. 111, 1872–1891 (2002)]) and are classed as ‘‘onset’’ and ‘‘offset’’ subtypes. A useful characteristic of landmarks is that their sequential combinatorial properties are limited: Given physiological constraints, only 44% of potential landmark pairs are legal. In this study, bigrams are used to quantitatively model the grammar of consonant landmarks. This work expands Liu’s method [S.A. Liu, J. Acoust. Soc. Am. 100, 3417–3430 (1996)], which detects the spectral abruptness of the speech signal, to a probabilistic framework, and then bigrams are applied to determine the most likely landmark sequences. Results tested on TIMIT are consistent across gender and several dialects of American English. Though lower thresholds than Liu’s are used to allow more insertions in detecting landmark candidates, the insertion errors are reduced by 46% due to bigram constraints. Many of the errors are systematic, reflecting variants such as flaps or syllabic nasals; they can be used to improve detection methods, and help us further understand acoustic variation in speech. [Work supported by NIH/NIDCD DC02978, DC00075, and T32DC00038.]
Investigating the consonant-vowel boundary: Perceptual contributions to sentence intelligibility
Authors: Fogerty Daniel , Kewley-Port Diane
Published: September 2008

Speech sounds divide into two broad categories: vowels and consonants. Vowels have a distinct perceptual advantage over consonants in determining sentence intelligibility [Kewley-Port, Burkle, and Lee, J. Acoust. Soc. Am. (in press)]. However, specifying the segmentation of consonants and vowels is problematic because of their extensive acoustic overlap due to coarticulation. The current study used TIMIT sentences to investigate perceptual contributions of consonants and vowels across systematic changes in the consonant-vowel (C-V) boundary. Sentences were presented to listeners with either consonant or vowel information replaced by noise. The C-V boundary was shifted by six specific proportions of the vowel, such that consonant duration increased while vowel duration decreased, yielding 12 different noise replacement conditions. The percent of words listeners repeated correctly was scored. A two-to-one vowel advantage for intelligibility at the traditional C-V boundary was confirmed. Initial results from ten listeners suggest that functions of the shifted C-V boundaries for consonants and vowels differentially contribute to intelligibility. Preliminary comparison of these functions suggests that both consonants and vowels contribute equally (50%) to intelligibility at about a 20% proportional boundary shift. Results will be interpreted in terms of phonological versus acoustic accounts of speech perception. (Supported by NIH)
Numerical study of the influence of roughness on coherence of signals reflected by the seafloor
Authors: Guillon Laurent , Holland Charles W.
Published: September 2008

The context of this work is the study of broadband acoustic signals reflected by the seafloor and recorded on a 15-hydrophone vertical array. Two different aspects of the coherence of these signals were examined: in the temporal domain with the maximum of the cross-correlation coefficients and in the frequency domain with the phase of the cross-spectra. In previous work, it was shown that these coherence parameters are sensitive to the geoacoustic nature of the seafloor which may open the door for a new method of probing the seabed properties on both large scales (i.e., deterministic) and small scales (i.e., statistical). The aspect examined in this present work is the influence of the seafloor roughness on the coherence. To do so, the time series are simulated by a three-step process: 1) computation of the plane-wave reflection coefficient, 2) computation of the spherical wave reflection coefficient by a numerical integration of the Sommerfeld integral, and 3) Inverse Fourier Transform. Using these simulations, with the measured data as a guide, the potential for detecting and extracting roughness estimates from the coherence in complex multilayered seabed environments is explored.
Response uncertainty using the stochastic finite-element method
Authors: Burkhardt John
Published: September 2008

Acoustic field predictions in shallow water are complicated by irregular bottom profiles and composition, sea-surface effects, as well as poorly characterized sound speed fields. To model the natural uncertainty of the problem, a variety of stochastic methods have been used. This presentation will consider the use of the stochastic finite-element method for the solution of the problem. The presented approach differs from previous spectral approaches in that the spatial problem, isolated by use of a polynomial chaos expansion, will be solved using the finite-element method, i.e., using a finite expansion of piecewise linear shape functions. Use of the finite-element method allows for the natural incorporation of irregular boundaries, as well as spatially and temporally dependent material properties. To demonstrate the technique, a one-dimensional elastic rod with spatially varying stiffness (sound speed) was considered. Specifically, the mean value of the rod stiffness and the standard deviation of the rod stiffness were varied in space. The technique was successfully implemented and confirmed using Monte Carlo techniques.
Uncertainty and coherence function for the scattering by a time-varying, two-dimensional, rough surface
Authors: Scharstein Robert W.
Published: September 2008

A perturbative analysis is used to construct and evaluate the mutual coherence function of the scattered field due to point-source excitation of a time-varying, rough surface. The wide-sense stationary (statistically homogeneous) rough surface is characterized by a simple time-space autocorrelation function that displays features of the moving sea surface. Doppler effects and evanescent wave contributions are explicit in the plane-wave spectral terms in the perturbation series, where the small parameter is the mean-square surface height.
Impact noise in a prefabricated custom modular condominium
Authors: Ermann Michael , Tew Gret , Golden Matthew V.
Published: September 2008

This line of inquiry aims to provide field-measured impact noise data for ‘‘box-on-box’’ construction, as was found in field tests. This type of construction holds the potential to increase impact isolation class (IIC) because the very process of building discrete boxes off site, transporting them, and stacking them on-site, establishes an airspace between the floor joist of one box and the ceiling joist of another. This partially decouples the floor and ceiling (a ridged connection remains at the perimeter of the boxes). A preliminary test of wood finished flooring yielded an FIIC of 42. Ongoing supplemental tests with floor underlayments and resilient seismic connections at the perimeter of the boxes will also be presented.