bluuebunny/crossref_metadata_2025 huggingface dataset:
Acoustic scattering of metallic springs in elastic medium
Authors: Zhang Jinmin , Varadan Vijay K. , Varadan Vasundara V.
Published: July 2012

In applications such as ultrasonic immersion testing, damping of flow noise in pipes and ducts and coating of acoustic control surfaces, it is very useful to reduce the reflection of acoustic waves from surfaces. Significant echo reduction can be achieved by addition of an anechoic coating. According to some general design guidelines for a single layer of homogenous coating, a single layer of 15 dB coating must be at least one wavelength thick. For absorbing acoustic waves of low frequencies, inhomogeneities such as metallic fillers, cavities and irregular metallic helices can be introduced into the lossy matrix to substantially enhance the acoustic absorption by acoustic scattering. In this presentation, the acoustic scattering of a single metallic spring embedded in an infinite elastic medium has been investigated using a boundary element formulation. The cross sections of the scattered longitudinal and shear waves have been calculated for various spring sizes, orientations and wave numbers. It will be shown that helical inclusions are more efficient in scattering longitudinal incident waves than spherical inclusions of equal volume. Steel helices are found to be more efficient in converting the incident longitudinal waves into shear waves than helical cavities of equal size.
Improved speech intelligibility in noise with a single-microphone noise reduction technique
Authors: Nilsson Michael J. , Fang Xiaoling , Ghent Robert M. , Murphy Patrick , Bray Victor H.
Published: July 2012

Reception threshold for sentences (RTS) as measured by a modified version of the HINT test [Nilsson et al., J. Acoust. Soc. Am. 95, 1085–1099 (1984)] were collected on 26 hearing-impaired listeners fit binaurally with digital hearing aids incorporating a nine-channel spectral subtraction technique of single-microphone noise reduction. Thresholds were measured in quiet and noise presented at a zero degree azimuth with the subjects listening unaided, aided without noise reduction, and aided with noise reduction and no modification to the gain function when noise reduction was activated. Additional data included soundfield thresholds as a frequency-dependent measure of the impact of the noise reduction algorithm on sensitivity to soft sounds. RTS in quiet reveal a significant effect of noise reduction [F(3,69)=46.9, p<0.01] with noise reduction increasing RTS 3.4 dB relative to no noise reduction, from 40.4 dB (A) to 43.8 dB (A). RTS in noise reveal a significant effect of noise reduction [F(3,69)=18.7, p<0.01] with noise reduction decreasing RTS 1.8 dB relative to no noise reduction, from 0.0 dB S/N to 1.8 dB S/N. Soundfield thresholds reveal a loss of sensitivity mainly in frequencies below 1000 Hz. The noise reduction algorithm will be reviewed and discussed.
Automatic speech recognition for mobile hand-held devices
Authors: Rose Richard , Narayanan Shrikanth , Parthasarathy S. , Rosenberg Aaron , Gajic Bojana
Published: July 2012

The implementation and evaluation of an automatic speech recognition (ASR) based task for a mobile, hand-held device is presented. The use of ASR on mobile devices will make environmental noise, and the design of transducers and compensation algorithms for dealing with noise, critical issues for the success of ASR based services on these devices. A description of the task is provided along with a set of compensation techniques that are used to compensate speaker independent hidden Markov models (HMMs) with respect to environment and transducer variability. Data were collected in a prototype application environment for a form-filling directory access task with speech and pen input and text output (48 subjects in an office environment; 21 in a cafeteria). A technique for combined environment/transducer compensation is presented for this task and is shown to significantly reduce the effects of environmental mismatch. The overall performance degradation was reduced from 41.7% to 10.4% for speech spoken through a far-field microphone in an office environment, and from 79.2% to 39.8% for the same transducer in a noisy cafeteria environment.
Soft-decision speech signal estimation
Authors: Krasny Leonid
Published: July 2012

Speech is often corrupted by background noise which degrades the performance of coding and recognition algorithms. This scenario is especially true in hands-free mobile communications where noise can be generated by the car engine, the road, and wind hitting the windows. It is essential to reduce the noise level without distorting the original speech signal. A well-known technique for solving this problem is spectral subtraction (SS), where the power spectrum of the noise is estimated when speech is absent and used to suppress noise when speech is present. Normally, a voice activity detector (VAD) is used to determine the presence of speech. In the conventional approach, VAD and SS algorithm act independently. Specifically, the speech signal is first detected by the VAD, and once the presence of the speech is detected, the SS algorithm is used to estimate the signal. Although each task can be optimized separately, the resulting algorithm is not optimal. In this paper, we present a novel approach for estimating the speech signal in which the detection and SS tasks are combined optimally. Our approach employs a soft-decision algorithm for detecting and estimating the speech signal, which leads to improved speech quality.
Pitch determination by fundamental-harmonic extraction using MUSIC algorithm
Authors: Murakami Takahiro , Ishida Yoshihisa
Published: July 2012

Short-term analysis pitch determination was executed by using multiple signal classification algorithm, which is an eigen-based subspace decomposition method proposed by Schmidt (Ph.D. thesis, Stanford University, 1981). The MUSIC spectrum, based on subspace principles, is sharply peaked at the frequencies of the sinusoidal components of speech signals without almost receiving the influence of the noise. Since the harmonic structure of the power spectrum of speech signals becomes unclear in a high-frequency domain, fundamental-harmonic extraction is performed using the band-limited MUSIC spectrum. The influence of the noise decreases further by limiting the frequency domain to be analyzed, and calculation time has been shortened greatly. The IDFT of the logarithmic MUSIC power spectrum exhibits a strong peak at the position equal to the pitch period like the cepstral method. Results of pitch determination for male and female Japanese vowels illustrate that the proposed method is more excellent than the cepstral method and can estimate the pitch frequency not being influenced by the noise.
Comparison between physical and auditory parametrization of speech corpus for the unit selection in text-to-speech synthesis
Authors: Tsuzaki Minoru
Published: July 2012

It is important to design an appropriate cost function to improve the quality of speech produced by a corpus-based text-to-speech (TTS) synthesis. Although the final product of the TTS system is evaluated perceptually, the definition of cost functions has to be based on the physical parameters of speech signal. And a cost function will become inappropriate if the discrepancy between the physical and the perceptual measures becomes critically large. The discrepancy could be caused by two main factors, i.e., (1) inappropriate selection of parameters, and (2) inappropriate scaling. One may judge that the selection of physical parameters is inappropriate if changes in a single physical parameter result in changes in multiple perceptual attributes. One may also judge that the scaling of physical dimension is inappropriate if the mapping function of the physical space to the perceptual space appears to be critically nonlinear. Aiming at a better definition of cost functions, a signal analysis based on the standard linear prediction coding (LPC) method and one based on the models of the peripheral auditory system (DSAM-AIM) are compared. The results of principal component analysis for Japanese speech databases of a corpus-based TTS system, CHATR, are presented.
Sound propagation using impulsive sources on the New Jersey Continental Shelf
Authors: Badiey Mohsen , Carey William M.
Published: July 2012

An acoustic experiment was performed on the New Jersey Continental Shelf in a well surveyed area with known bottom and sub-bottom properties. The measurement system was a calibrated vertical array 14–18 km from a known geological feature. Calibrated airgun shots (AG) and special small omni-directional explosives shots (SO) were used to generate impulsive sounds monitored by a calibrated source hydrophone. The repeatability of the SO and AG shots was excellent. The sea state during this experiment was between 4 and 6 on the Beaufort scale and the ship was held in position on either side of the geological feature. The sound velocity profiles were stable over the measurement period due to the fully developed seas. A mixed layer depth of approximately 20–30 m was followed by a steep thermocline to a depth of 48 m with a gradient of approximately 0.5 sec and a relatively isothermal layer to the bottom which ranges between 70 m at the source location and 80 m at the receiving array. Measurements received on the vertical array were variable but over short periods of time remarkably consistent. Time of flight enabled the separation of the first three modes whose shape and arrival times were to agree with calculations. Comparisons of the arrival structure changes due to the feature and estimates of the backscatter from this feature are presented in relation to the mode arrival statistics for different groups of shots.
Time variability of acoustic signals in a benign shallow-water area
Authors: Nielsen Peter L. , Siderius Martin , Jensen Finn B.
Published: July 2012

SACLANTCEN conducted the ADVENT’99 experiment in the Strait of Sicily, Mediterranean, to assess the time variability of broadband acoustic signals in a benign shallow-water region. Broadband acoustic signals (200–3800 Hz) were transmitted for up to 18 h over fixed paths of 2, 5, and 10 km. Dense sampling of the environment was performed including a sound-speed section every hour along the acoustic tracks. Band-averaged transmission loss is stable with time, which agrees well with modeling results. The Bartlett processor was applied to correlate the acoustic data. The correlation time varies from several hours at low frequency and short range to a few minutes at high frequency and longer ranges. Range- and time-dependent propagation modeling of the acoustic signals (200–800 Hz) shows behavior similar to the data. Particularly, the correlation time decreases abruptly from several hours to less than 1 hour at a frequency around 700 Hz for the 10-km data. This effect of the time-varying waveguide on the acoustic signals is correctly predicted by the propagation model. Severe problems using the acoustic signals coherently persist above a few hundred Hz and at ranges beyond a few kilometers, although the experiment was conducted under favorable conditions. The experimental data and the simulations illustrate the complexity of sound propagation in shallow water.
Challenges and more in the acoustic design of an attraction
Authors: Shaw Neil A. , Van den Broeck Ray
Published: July 2012

Building a theme park from scratch involves not only providing a ‘‘theme’’ for the attractions contained therein, but also integrating the acoustic control and enhancement measures so that they work with, and not against, the goal of the attraction, which is the suspension of disbelief, at least for the age group to which the attraction is designed. The challenges encountered by the authors, the analysis of the same, as well as the proposed solutions, in the acoustic design for an enclosed attraction that includes hands-on interactive activities, live performances, and a roller coaster will be reviewed. Field work to determine potential mitigation paths will also be discussed.
Knott’s Camp Snoopy at the Mall of America: A theme park under glass, revisited
Authors: Thorburn Steven J.
Published: July 2012

Over ten years ago, the design of a special theme park was born. The plans for Knott’s Camp Snoopy at the Mall of America in Minneapolis, Minnesota would eventually grow into a popular tourist destination, but there were initial concerns about acoustics. The owners were familiar with the high noise levels of other indoor amusement parks and wanted to avoid that same situation. Having just completed the West Edmonton Mall, where many complaints were received about noise levels and guest visits were very short, it was very important to the owners that the design would prevent problems associated with noise. To do this, an acoustical consulting engineering firm was brought in. In this paper, the original acoustical design goals and solutions will be reviewed and compared against the final constructed project and its success over the first decade.
Noise abatement strategies and modeling approaches for outdoor attractions at theme parks
Authors: Menge Christopher W.
Published: July 2012

Recent experience in noise abatement approaches and results for a new theme park are presented. Significant contributions from different source components on roller coasters lead to different abatement strategies taken for each. Measured source data will be presented. The relative performance of abatement applied separately to the source and sound path will be discussed, as well as the effects of source directivity. Source and propagation modeling approaches are presented, including how a customized version of a recently developed national noise model for highway traffic was applied for roller coasters and other sources.
An autonomous acoustic recorder using a directional sensor for locating calling bowhead whales
Authors: Norman Robert G. , Greene Charles R.
Published: July 2012

A new directional acoustic recorder has been developed to assess the movement and distribution of vocalizing marine mammals. Called DASAR (Directional Autonomous Seafloor Acoustic Recorder), the instrument incorporates an omnidirectional and two orthogonal horizontal sensors from a DIFAR (DIrectional Frequency and Recording) sonobuoy, as well as a magnetic compass. With appropriate processing, the horizontal directions can be computed for received sounds. The acoustic bandwidth is 1000 Hz, appropriate for bowhead calls, and recording is continuous for 45 days. DASARs were deployed in a hexagonal array seaward of an artificial gravel island recently constructed for oil and gas production northwest of Prudhoe Bay, Alaska. Operation was during the fall migration of bowhead whales past the island. The objective was to relate calling-whale locations to the character and levels of sounds emanating from the island and from nearby tugs and barges. [Work supported by BP Exploration (Alaska).]
A digital acoustic recording tag for measuring the response of marine mammals to sound
Authors: Johnson Mark , Tyack Peter , Nowacek Douglas , Shorter Alex
Published: July 2012

There is growing concern about the effect on marine mammals of underwater sound sources such as vessels, industrial equipment, and sonar. In order to quantify the reaction of these animals to sound, controlled exposure experiments have been attempted using surface observations or vocal monitoring to estimate response. However, the short surface time of most marine mammals, and their often unknown vocalization rates, limit the effectiveness of such experiments. To address these problems, a small digital recording tag, the DTAG, has been developed capable of simultaneously sampling the acoustic environment of the host animal, together with its orientation, heading, and depth. The tag has a 400-Mbyte memory array sufficient to record audio and sensor signals for several hours. The tag is encapsulated in a plastic resin and can operate at a water depth up to 3000 m. Programing and data off-load are accomplished with an infrared communications link to a personal computer. The DTAG has been deployed on northern right whales and sperm whales using suction cup attachments. Delivery was by means of a cantilevered pole from a small boat. The resulting data indicate a number of behavioral response metrics as well as new results on dive behavior and vocalization rates.
Detection of harbor porpoises and white whales (Beluga) sound using the high frequency bioacoustics and their echo structures
Authors: Mutlu Erhan
Published: July 2012

Acquisition of sounds emitted by marine mammals (porpoise and beluga) living in the Black Sea was made using active echosounder (BioSonics Model 102) with down-looking 120- and 200-kHz transducers. The sounds of the mammals displayed different structures not similar to each other. The individual amplitude of the porpoise sound was significantly discriminated and varied between −55 and −62 dB at 200 kHz. The echoes were characterized with decreasing amplitude from the deep water to the surface. It looked like vertical lineations. The sound scattering was fluctuated in a range of 10–20 dB. Sound emitted by beluga displayed a structure like rakes of a comb. The echo was captured first very weak at 120 kHz and then strong at 200 kHz. The volume scattering changed within a range of 60 and −70 dB at 200 kHz. While the whale was close to the transducer, it emitted the sound every 300 s. It sometimes gave sound in different patterns from some reason. As we got away from the whale, while on the way, we detected the sound at 10 s. Each lineation of the comb-rake structure appeared in a certain time delay and the strength decreased toward the surface.
Using MODAS to derive subsurface sound speed estimates from satellite observations
Authors: Barron Charlie N. , Fox Daniel N. , Holmes Eleanor
Published: July 2012

The Modular Ocean Data Assimilation System (MODAS) provides a set of tools for analysis and manipulation of ocean data. One of its functions is the estimation of subsurface temperature and salinity fields from satellite observations of height deviation and temperature. A two-dimensional optimal interpolation of the satellite data produces fields of surface height anomaly and temperature. These data are provided to the MODAS synthetic profile routine to predict subsurface temperature and salinity fields compatible with the surface observations and local climatological statistics. Available in situ measurements may be assimilated at this stage. A sound speed field calculated from the temperature and salinity estimates provides the basis for additional acoustical applications. MODAS products, evaluated by comparison with finely sampled surveys, show significant improvement over estimates from static climatologies. [This work was funded by the Space and Naval Warfare Systems Command (SPAWAR) and the ONR through the Advanced Data Fusion Center Support and the On-Scene Tactical Ocean Forecast Capability projects (SPAWAR, Program Element 603207N) and the MODAS Improvements and the Prediction of Coastal Buoyancy Jets projects (ONR, Program Element 602435N).]
The dependence of low-frequency underwater surface scattering on remotely sensed oceanographic variables
Authors: Gauss Roger C. , Thorsos Eric I. , Henyey Frank S. , Fialkowski Joseph M.
Published: July 2012

Recent low-frequency (<1500 Hz) underwater acoustic measurements have demonstrated that scattering processes at the air–sea interface depend primarily on the wind speed and the surface wave spectrum. The scattering strength from the rough interface is proportional to the spectral density at the Bragg wavelength with modifications due to tilt and modulation by longer waves. These modifications are accounted for in the small slope approximation. When wave breaking becomes significant, rough interface scattering is augmented by bubble cloud scattering, which depends primarily on wind speed. In this regime, bubble cloud scattering dominates at low grazing angles, and rough interface scattering dominates at high grazing angles. A physics-based empirical model is used to describe bubble scattering. The mean-frequency-shift characteristics of acoustic signals scattered from both the moving sea surface and bubble clouds have been successfully modeled given the 2-D surface wave spectrum. These scattering-strength and frequency-shift models are used to explore the sensitivity of low-frequency scattering to environmental variables obtainable by remote sensing. [Work supported by ONR.]
The estimation of acoustic properties of internal waves using synthetic aperture radar (SAR)
Authors: Small Justin , Prior Mark
Published: July 2012

This paper looks at some methods relevant to the assessment of the acoustic properties of internal waves from satellite data. The currents in these internal waves strain the sea surface leading to signatures, which can be imaged by instruments such as radar, photography, and SAR. Two lines of approach for estimating internal wave amplitudes from SAR will be illustrated. The first approach uses length-scale information on SAR (separation of waves, width of waves), combined with some knowledge of the stratification (from a database or survey) and a solitary internal wave or undular bore model. The second approach uses radar imaging models to investigate the strength of current which would lead to the observed signatures in the given wind conditions. One important acoustic property of internal waves is the influence on surface duct propagation due to the large measured excursions of the mixed layer. Examples will be shown of the effects on surface duct propagation for different frequencies and scenarios using experimental data taken (1) off the northwest of the UK; (2) in the central Bay of Biscay; and (3) on the Portuguese coast.
The multiple source feedback of supersonic jet screech, III: Space-time diagrams applied to the helical mode
Authors: Powell Alan
Published: July 2012

The foregoing feedback theory applies directly to the sinuous and toroidal instability modes of supersonic jets, but needs modification for the helical C mode since the acoustic feedback to the nozzle must at the axial trace velocity of the resultant helical sound field (Powell, Umeda, and Ishii, 1990). Here the space-time diagram is equivalent to an instantaneous map of the positions of the helical instability and acoustic wave on the cylindrical bounding surface of the jet, one period being replaced by one circumference. The instantaneous space-time (sound) pressure measurements of Westley and Woolley (1975) are interpreted in this way. The acoustic feedback trace velocity is found to be approximately sonic. The circumferential velocity (=πf×diameter source orbit diameter), in a plane normal to the jet axis is inferred to be supersonic. The phase velocity of the helical wave along the curved surface is then found to be subsonic rather than sonic as originally assumed in the tentative theory. A simple numerical simulation (including the reflection from the flange) reproduces the main features of the complex map of instantaneous pressure, including the prominent standing waves.
Nonprofits and noise education, the experience of the Noise Pollution Clearinghouse
Authors: Blomberg Leslie D.
Published: July 2012

During its four-year history, the Noise Pollution Clearinghouse has become a national leader in educating the public about noise and its effects. This paper explains the work of the Noise Pollution Clearinghouse, the reasons for its success, and the unmet needs and opportunities that remain. Particular attention is paid to the potential for nonprofits and professional societies to be leaders in noise education.
Recent work and new axes of research for m
o
d
a
l
y
s, a sound synthesis program based on modal representation
Authors: Vergez Cristophe , Bensoam Joel , Misdariis Nicolas , Causse Rene
Published: July 2012

modalys, developed at Ircam after 10 years for research and musical applications, is based on modal representation of vibrating objects, mechanical or acoustical. Modal data (natural frequencies, damping factors, and deflection shapes) are obtained from: (1) analytic modeling for simple structures as ideal string, cylindrical or conical open/closed tubes, rectangular plate with a simply supported boundary condition or violin bridge, ...); (2) experimental measurements for real structures with complex shapes (violin body, soundboard of piano, flute,...); and (3) finally by numerical methods (finite-element methods) which allow the creation of any arbitrary virtual shapes. The vibrating object involved is generally a set of simple and linear substructures. Nonlinearities, when they exist, are introduced in the coupling of substructures, for example reed and tube for the clarinet. Dissipation plays a critical role in the perception of some vibrating musical objects. The type of damping used in our modal representation is referred to as proportional damping (dissipation phenomena are in this case constant or proportional to the square of frequency). However, this type of damping may not apply in any case to the real objects. In this presentation, some sound illustrations on recent results of research and application development with modalys will be presented and discussed.
A study of brass instrument ‘‘lipping’’ using an artificial mouth
Authors: Neal Mark , Barnes Francis H. , Campbell D. Murray , Greated Clive A. , Cullen John S. , Gilbert Joel
Published: July 2012

Sound is generated in a brass wind instrument as a result of aeroelastic coupling between the mechanical vibratory system formed by the lips and the acoustical vibratory system of the air column. The sounding frequency depends on the parameters of both vibratory systems, and is in general not exactly equal to one of the acoustical resonance frequencies of the air column. A skilled brass instrument player can vary the method of blowing to pull the note above or below an acoustic resonance; this technique is known as ‘‘lipping.’’ In order to obtain a clearer understanding of the lipping mechanism, an artificial mouth has been used to drive a trombone mouthpiece coupled to a cylindrical tube of variable length. The phase relationship between mouthpiece pressure and lip opening has been determined for near-threshold auto-oscillation over a range of air column lengths and embouchure adjustments, corresponding to lipping both above and below the acoustic resonance. Implications for the classification of the lip reed mechanism are discussed.
Sound attenuation in the critical region of the ionic binary mixture ethylammonium nitrate/n-octanol
Authors: Mirzaev Sirojiddin Z. , Kaatze Udo
Published: July 2012

Ultrasonic attenuation spectra (100 kHz<f<4 MHz) of the ionic mixture ethylammonium nitrate/n-octanol of critical composition and also of the ethylammonium nitrate itself (100 kHz <f< 400 MHz) are discussed at different temperatures. The measured spectra are evaluated in terms of the Bhattacharjee–Ferrell dynamic scaling theory. Using literature data for the amplitude of the fluctuation correlation length, the background and critical part of the heat capacity, the sonic attenuation spectrum as predicted by the Bhattacharjee-Ferrell model has been calculated for the critical mixture at the critical temperature. Again following this theoretical model, the contribution due to concentration fluctuations has been subtracted from the measured spectra at the temperatures of measurement and also subtracted has been the high-frequency asymptotic background contribution. The measured data are consistent with the Bhattacharjee–Ferrell theory of critical concentration fluctuations but show an additional frequency dependence. [Work supported by the Volkswagen-Stiftung, Hannover, Germany.]
Distance localization using nonindividualized head-related transfer functions
Authors: Zahorik Pavel
Published: July 2012

Past research has suggested that virtual auditory displays are most accurate when implemented using individualized head-related transfer functions (HRTFs). Directional localization accuracy, for example, is often substantially degraded when nonindividualized HRTFs are used, due to the resulting distortion of spectral cues to sound source direction. Although equally important for accurate spatial representation, little is known about distance localization accuracy in virtual auditory displays, and specifically, whether individualized HRTFs are necessary for accurate distance localization. Here, distance localization accuracy was assessed for six listeners, using both individualized and nonindividualized HRTFs in separate blocks of trials. The virtual sound sources were produced from special HRTF measurements (source distance ranging from 0.3 to 13.8 m) that also contained the acoustic transfer characteristics of a semireverberant room environment. This provided listeners with additional acoustic cues to source distance, including the ratio of direct-to-reverberant sound energy. Since no significant differences were observed in listener’s judgments of apparent sound source distance in the two stimulus conditions, it is concluded that distance localization accuracy, and degree of sound source externalization, do not depend on the use of individualized HRTFs when natural reverberation is available to the listener. [Work supported by NIH-NIDCD, NASA, and NIH-NEI.]
AM depth versus degree of amplitude fluctuation: Implementation error, adjustment, and implications
Authors: Vassilakis Pantelis
Published: July 2012

AM depth values are often used in signal processing as a measure of amplitude fluctuation. In such cases, AM implementation produces an error reflecting the nonlinear relationship between the presumed and applied AM depths. A distinction has been made between two concepts of AM depth: (a) ‘‘spectral AM-depth,’’ referring to the relationship between AM depth and degree of spectral energy spread and (b) ‘‘envelope AM-depth,’’ referring to the relationship between AM depth and degree of amplitude fluctuation. The usual AM implementation equation represents consistently only the ‘‘spectral AM-depth.’’ It is shown that, in order to apply an intended ‘‘envelope AM-depth,’’ an adjusted AM depth coefficient must be inserted in this equation. The adjustment affects studies where changes in the degree of amplitude fluctuation have been correlated to changes in some physical, physiological or perceptual measurement. In general, all functions derived from the results of studies based on the miscalculation of amplitude fluctuation have been distorted by a factor proportional to the identified error. If and how this distortion influenced the interpretation of the provided functions will vary according to the importance and meaning attributed to each specific functions shape. Specific examples and general implications are discussed.
Acoustic atomic force microscopy applications in nanotechnology
Authors: Tittmann Bernhard R. , Gao Weilou , Du Jikai
Published: July 2012

In this work, acoustic atomic force microscopy is presented for nanometer material characterization. A transducer with a frequency range from 0.1 kHz to 1 MHz is typically fixed under a sample. The resonant frequency for the composite system is determined experimentally. The amplitude and phase images of the vibrating composite are obtained simultaneously as well as surface topography. Plasm spray drying process is studied by use of the acoustic AFM. Enhanced contrast in amplitude images shows unique capability to determine the plasm distribution in Alumina matrix. Further study is conducted on the contrast mechanism with a sample comprised of three different materials: Si3N4, Cu, and steel. Theoretical analysis based on finite-element methods (FEM) and Johnson, Kendall, Roberts (JKR) model is conducted to simulate the system and determine the resonant frequency, amplitude, and phase. The results of the numerical simulations compare favorably with the experiment results. Furthermore, an amplitude-based method is proposed for estimating the material modulus through a semitheoretical, seminumerical process. The method shows higher sensitivity than the resonant frequency-based method. Also, microelectronic devices are studied for the analysis of diffusion.
Development of a SAW microgas sensor for monitoring SO2 gas
Authors: Roh Yongrae , Lee Youngjin , Kim Hakbong
Published: July 2012

We have developed a new type SO2 gas sensor applying a particular inorganic thin film on SAW devices. The sensor consists of twin SAW oscillators of the center frequency of 54 MHz fabricated on LiTaO3 piezoelectric single crystals. One delay line of the sensor is coated with a CdS thin film that selectively adsorbs and desorbs SO2, while the other is uncoated for use as a stable reference. Deposition of CdS thin film has been carried out by the spray pyrolysis method using an ultrasonic nozzle. Mass loading induced by the SO2 gas adsorbed onto the CdS film results in corresponding frequency shifts directly proportional to the gas concentration. The relative change in the frequency of the two oscillators is monitored with a digital signal processing circuit. SO2 sensor properties investigated include selectivity, sensitivity, response time, and repeatability. The sensor shows promising performance as a microsensing tool and is capable of measuring concentrations in air less than 200 parts per billion of SO2.
Channel interaction and speech processing strategies for cochlear implants
Authors: Stickney Ginger S. , Loizou Philip , Mishra Lakshmi Narayan , Assmann Peter F. , Shannon Robert V. , Opie Jane M.
Published: July 2012

A limitation of multichannel cochlear implants is that electrical channel interactions can arise when multiple electrodes are stimulated simultaneously. Channel interaction can be reduced by decreasing current levels delivered to each electrode, through improved electrode positioning and design, or by using speech processing strategies that stimulate electrodes sequentially. The goal of the present study was to examine the relationship between channel interaction and speech recognition for several speech strategies varying in the number of electrodes stimulated simultaneously. Based on previous research, susceptibility to channel interaction is expected to vary as a function of electrode configuration. Thus patients with strong susceptibility to channel interaction will produce higher speech recognition scores for sequential as opposed to simultaneous speech strategies. Speech strategy performance was evaluated with acute listening trials where patients identified consonants and vowels with each of the following speech strategies, listed in order from sequential stimulation to fully simultaneous stimulation: Continuous Interleaved Sampler (CIS), Paired Pulsatile Sampler (PPS), Hybrid Analog Pulsatile (HAP), and Simultaneous Analog Stimulation (SAS). Preliminary results showed that a greater degree of channel interaction was associated with the highest speech recognition scores for CIS, midrange scores for HAP, and the lowest scores for SAS. [Work supported by NIH.]
Feature extraction in human speech recognition
Authors: Allen J. B. , Furst M. , Saul L. , Rahim M.
Published: July 2012

In 1955 George Miller and Patricia Nicely explored the human speech code and discovered that a five-channel feature code is used to extract phonemes from speech. Their mutual information analysis of the confusion matrices of filtered noisy speech showed that the five channels were virtually independent of each other, accounting for the robustness of speech communication quantified by Miller, Heise, and Lichten in 1951. While the 1955 analysis was a major step forward, it did not lead to practical results, since there are no known methods for extracting these features from the speech. This weakness is the basis of the present research program, to define the signal processing required to robustly extract features from narrow bands of speech. Our emphasis is to work in narrow bands, starting from critical bands. In this paper we discuss the first step of this process, which is to measure (a) the jnd for the detection of speech in noise and (b) the jnd for residue pitch given narrow bands of speech in noise. We have been investigating these features with the ultimate goal of trying to build a speech recognition machine based on distinctive feature signal processing.
Vowel perception in varied symmetric CVC contexts
Authors: Talley Jim
Published: July 2012

In the three-decade-long debate over static versus dynamic specification of vowels, perceptual studies in which subjects are tasked with identifying naturally spoken vowels under various ablation conditions have been a mainstay. While not directly producing an understanding of how humans go about recognizing this major subclass of phones, this type of study [e.g., Strange, Jenkins, and Johnson, J. Acoust. Soc. Am. 74, 695–705 (1983)] has provided compelling results which must be accounted for in any successful theory of vowel perception. This paper presents results from yet another perceptual study of human vowel identification under ablation conditions. This study uses CVC syllables spoken rapidly by three male speakers in a carrier sentence. Syllables consist of ten American English vowels in each of four consonantal contexts (b—b, d—d, g—g, and h—d). The conditions studied are silent centers (SC), centers only (CO), and the control condition (full). A very robust hierarchy of full>CO>SC is found. Consonantal contexts also have a clear ordering (h—d>b—b>d—d>g—g) with respect to the ease with which they are perceived. Interesting interactions between vowels and their contexts are also evident.
Vowel formant discrimination in natural speech
Authors: Liu Chang , Kewley-Port Diane
Published: July 2012

Vowel formant discrimination in syllables, phrases, and sentences was measured for nearly natural speech resynthesized by straight [Kawahara et al., Speech Commun. 187–207 (1999)]. The purpose of experiment 1 was to validate the use of straight to modify formants by comparing vowel discrimination in formant-synthesized and -resynthesized isolated vowels. The resulting DLs for F1 and F2, calculated in barks, were the same as previously reported. In experiment 2, DLs were measured in natural speech for F1 and F2 for the vowels /ih, eh, uh, ae/. Different phonetic contexts included syllables, phrases, and sentences, as well as sentences with the addition of an identification task. Results showed that longer phonetic context was not a strong factor in degrading DLs, although performance for syllables was significantly better than for sentences. The addition of the identification task did not have a significant effect on vowel discrimination. In comparison to the formant-synthesized speech in Kewley-Port and Zheng [J. Acoust. Soc. Am. 106, 2945–2958 (1999)], vowel discrimination was similar for syllables, but significantly degraded for more complex contexts. However, the magnitude of the results suggests that vowel formant discrimination is similar for synthetic and natural speech in a variety of phonetic contexts. [Work supported by NIHDCD-02229.]
Time-reversal pulse compression in shallow water with a single-element source–receiver
Authors: Gaumond Charles F. , Fromm David
Published: July 2012

A time-reversing, full-water-column, source–receiver array has been previously demonstrated to focus and time compress a field at the position of the source through the use of all the propagating modes of the water column [W. S. Hodgkiss et al., J. Acoust. Soc. Am. 105, 1597–1604 (1999)]. Simulations are presented that show the limitations of using a single-element echo repeater instead of a full-water-column array. The system consists of a source located at the center of a vertical line array (VLA) and a distant, single-element echo repeater (SEER). The echo field produced by the SEER has very limited vertical extent of time compression at the range of the source and VLA. This vertical extent can be extended with the use of received signals from a varying number of adjacent hydrophones. The tradeoffs involved with extending the compression are shown. The simulations are generated using RAM in a simple downward-refracting shallow-water environment. [Work supported by ONR.]
Robust diagnosis for the focal stability in time-reversal acoustics
Authors: Song H. C. , Kuperman W. A. , Akal T. , Hodgkiss W. S. , Kim S. , Edelmann G.
Published: July 2012

For a time-independent medium, one can use stored probe pulses to focus on a specific location in time-reversal acoustics. However, the temporal variability of the ocean is expected to limit such a procedure. In the earlier low-frequency (445 Hz) experiments, we found that probe pulses up to 1 week old still produced a significant focus at the original probe source location. In the recent high-frequency (3500 Hz) experiments, however, stability of the focus was limited to less than an hour or even a few minutes, depending upon the oceanographic variability since the acoustic wavelength is much smaller than the one for 445 Hz. A quick and robust diagnosis for the focal stability appears to be the modal group velocity versus phase velocity (gp curves) derived from the sound-speed profile, where the phase speed is directly related to the mode number and the slopes of the curves separate different groups of modes. In this paper, we investigate the stability and scintillation of the focus observed during the May 2000 time-reversal experiment using the gp curves.
Steering nulls in TRM with distortionless response at focal location—Theory and simulation
Authors: Kim J. S. , Kuperman W. A. , Song H. C.
Published: July 2012

The time-reversal mirror uses the received signal from a probe source to refocus the signal at the probe source location by back-propagating the time-reversed signal. In this study, a method to place a null at an arbitrary position while maintaining a distortionless response at the probe source location, assuming that the transfer function is known at that point, is proposed and demonstrated via simulation in Pekeris ocean waveguide. It is further shown that the null can be placed in the vicinity of probe source at the same depth using the theory on the invariants of ocean waveguide, without any environmental information. The approach is based on the adaptive weighting, of which formulation is a classical minimization problem with a constraint known as the minimum variance with distortionless response. The proper weighting for an array is applied before backpropagation. Finally, it is demonstrated that the null can be placed at the probe source location.
Guidelines for underwater acoustic data telemetry system design
Authors: Edelson Geoffrey , Menas Francine , Richman Michael
Published: July 2012

The use of acoustic communications in any number of projected applications requires trade-offs within the realm of technologies currently available and those projected for the future. Analyses must be performed to predict the performance, strengths, and limitations of competing approaches. Because underwater acoustic data telemetry can be viewed as a union of sonar and communications, it suffers from the same environmental degradation as sonar and communications systems. Efforts undertaken to date have shown that system performance is dictated by received signal-to-noise ratio and multipath complexity, spread, and rate-of-change. Comparisons between different signaling technologies can be made using a variety of measures of effectiveness. These metrics provide a detailed structure for the evaluation of the performance envelope as a function of environmental and platform conditions. We have compiled published results on a large selection of modems in terms of range-rate product, bandwidth efficiency, range, frequency, and power. Based on these compiled results and on simple physical models, design rules-of-thumb have been derived to ensure that proposed system designs have communication requirements that are reasonable and robust. While these rules may seem somewhat conservative, the sample points reported in the literature may not have been achieved reliably or more than once.
Evaluating a method for field measurement of enclosure noise reduction
Authors: Sachs Felix Z. , Byrne David C.
Published: July 2012

Measurement of noise reduction for sound isolating enclosures in situ is often complicated by limited access to all enclosure surfaces and indeterminate reverberation conditions. Both laboratory and field noise reduction measurements were made on an audiometric enclosure. The laboratory measurements followed ASTM method E596 using a reverberation chamber. The field method consisted of placing a ‘‘calibrated’’ loudspeaker at 1 m from the enclosure surface, applying a broadband signal and measuring 1 mm from the enclosure surface as well as inside the enclosure. The speaker calibration consisted of mapping the sound levels at 1 m in front of the speaker cone in an anechoic chamber. The field noise reduction is estimated using the inside and outside sound measurements. Since the loudspeaker sound field is localized, a surface area adjustment for the field noise reduction is required. The adjustment factor was derived from the E596 noise reductions. The factor is shown to be related to the loudspeaker calibration. [The opinions or assertions contained herein are the views of the authors and are not to be construed as official or as reflecting the views of the Department of the Army or the Department of Defense.]
A three-dimensional coherent Doppler profiler
Authors: Zedel Len , Hay Alex
Published: July 2012

A 1.7-MHz coherent Doppler sonar system has been designed for near-shore environments that provides three-dimensional velocity profiles over an O (0.5-m) depth range. The use of a bistatic geometry allows three independent components of velocity to be measured simultaneously in time and coincident in space. The system was calibrated in a tow-tank facility for velocities up to 2 m/s. For multiple pulse-pair ensemble-averaged velocity estimates generated at a rate of 30 profiles/s in 0.7-cm-depth bins, vertical velocity accuracies of the order 1 cm/s are achieved. For the horizontal components (based on bistatic measurements), standard deviation in velocity estimates of about 5% is realized with absolute accuracies of about 2%. In the configuration as tested, flow disturbance around the instrument introduces a bias of about 5%: this disturbance is modeled using potential flow past a cylinder. The disturbance errors could be corrected to the free-stream velocities using the present data set. Such a correction would not, however, reflect the actual water speed at the point of measurement. [This research was funded by the Office of Naval Research Coastal Sciences Program, and the Natural Sciences and Engineering Research Council of Canada.]
Musical instruments of antiquity as illustrated in The Adventures of Asterix the Gaul
Authors: Russell Daniel A.
Published: July 2012

The Adventures of Asterix the Gaul, a series of books written by Goscinny and illustrated by Uderzo, between 1960 and 1999, have received much acclaim for the attention to detail in their drawings of ancient architecture and civilization. Equally interesting to a musical acoustician are the illustrations of musical instruments (including carnyx, buccina, lur, bagpipe, harp, lyre, pipes, drums, and singing) used by ancient Romans, Greeks, Egyptians, and Gauls. This talk will compare Uderzo’s illustrations to photographs of period instruments and comment on their acoustic qualities, performance techniques, and the roles they played in their respective societies, both in real history and as experienced by Asterix and his friends.