bluuebunny/crossref_metadata_2025 huggingface dataset:
Phonetic context effects in adult listeners with cochlear implants
Authors: Aravamudhan Radhika , Lotto Andrew J.
Published: October 2013

From previous studies it is known that normal-hearing (NH) listeners have the ability to compensate for the acoustic variability present in speech through context-dependent perception of speech sounds. One question of practical and theoretical interest is whether listeners with cochlear implants (CI) also show context-dependent speech perception. Because of the lack of spectral resolution in the input, the representation of speech for CI listeners may differ from NH listeners, which may interfere with perceptual compensation. In a test of this prediction, adult postlingually deafened CI listeners did not demonstrate the contrastive context effects elicited from NH listeners for either /da/–/ga/ targets and /al/–/ar/ contexts or V targets and /b—b/–/d—d/ contexts. In contrast, as predicted by the good temporal resolution of the CI signal, CI listeners showed normal effects of vowel length on preceding glide-stop categorization. CI simulations with NH listeners were also performed for some of these context effects. The results support the view that spectral content of contexts largely determines their effect on target sounds, and that phonetic content does not play an essential role. In addition, the findings predict that CI listeners will have particular difficulty with heavily coarticulated speech. [Work sponsored by NIH and NSF.]
Speech production in experienced cochlear implant users undergoing short-term auditory deprivation
Authors: Greenman Geoffrey , Tjaden Kris , Kozak Alexa T.
Published: October 2013

This study examined the effect of short-term auditory deprivation on the speech production of five postlingually deafened women, all of whom were experienced cochlear implant users. Each cochlear implant user, as well as age and gender matched control speakers, produced CVC target words embedded in a reading passage. Speech samples for the deafened adults were collected on two separate occasions. First, the speakers were recorded after wearing their speech processor consistently for at least two to three hours prior to recording (implant ‘‘ON’’). The second recording occurred when the speakers had their speech processors turned off for approximately ten to twelve hours prior to recording (implant ‘‘OFF’’). Acoustic measures, including fundamental frequency (F0), the first (F1) and second (F2) formants of the vowels, vowel space area, vowel duration, spectral moments of the consonants, as well as utterance duration and sound pressure level (SPL) across the entire utterance were analyzed in both speaking conditions. For each implant speaker, acoustic measures will be compared across implant ‘‘ON’’ and implant ‘‘OFF’’ speaking conditions, and will also be compared to data obtained from normal hearing speakers.
Prediction of vowel intelligibility based on acoustic characteristics in speakers with Parkinson’s disease
Authors: Bunton Kate
Published: October 2013

This study investigated the contribution of acoustic features produced by three vowel contrasts (front–back, high–low, tense–lax) to the intelligibility of the contrast and overall speech intelligibility in speakers with Parkinson’s disease. Multiple regression analyses were used to determine which acoustic measures or combination of measures accounted significantly for the variance in the intelligibility measures. The predictor variables in the multivariate analysis included acoustic measures thought to represent essential perceptual cues to signal the vowel contrast. These findings are discussed in relation to specific areas of production deficiency that are consistent with Parkinson’s disease. [Work supported by NIH R03 DC005902.]
Source location in an urban setting using time reversal with small sensor arrays
Authors: Todaro Mark E.
Published: October 2013

The finite-difference, time-domain (FDTD) method was used with time reversal to explore acoustic pulse source location in an urban setting with the source outside the array of sensors. The FDTD program simulates two-dimensional sound propagation from an omnidirectional point source, including reflection and diffraction from buildings. Pressure versus time is recorded at various hypothetical sensor locations. During time-reversal playback, each sensor becomes an omnidirectional point source generating a time-reversed version of the pressure it received. A number of sensor arrays were modeled in various urban settings to identify the limitations of using time reversal to locate a non-line-of-sight source outside the array. In general, as the number of sensors and spatial extent of the array increases, the ability to locate the source increases as well. A 19-sensor hexagonal array 8 m across, for example, is capable of locating a source on the opposite side of a building. Two small four-sensor arrays 10 m from each other are adequate for around-the-corner source location. A single small array of three or four sensors spaced 1 to 2 m apart was generally insufficient to locate the source.
Bistatic clutter statistics in underwater acoustics
Authors: La Cour Brian
Published: October 2013

For the case of a spatially separated (i.e., bistatic) source-receiver pair, the joint distribution of the time delay, bearing, and Doppler shift of acoustic clutter is derived. The clutter is assumed to be spatially uniform with a fluctuating velocity that is Gaussian distributed with zero mean and components that are uncorrelated and of equal variance. For the Doppler distribution, the source and receiver are assumed, in general, to be moving relative to one another. Unlike the monostatic case, the time delay and bearing are found to be statistically dependent. Conditioned on the time delay, the distribution in bearing exhibits two peaks appearing to either side of the source direction for times less than three direct-blast times. Later times show a single peak in the direction of the source which diminishes in relative amplitude with time. The Doppler shift distribution, conditioned on time delay and bearing, is derived under the assumption that the clutter velocity dispersion is much smaller than the speed of sound. The statistical distribution is then related to bistatic reverberation, and a qualitative comparison is made between measured bistatic data and the derived statistical distributions.
Analysis of variable length data records for generating spectrally decomposable target-specific features
Authors: Supriya M.H. , Saseendran Pillai P.R.
Published: October 2013

The instantaneous noise data waveforms generated by surface crafts vary widely, and it is hardly possible to interpret them into a pattern that indicates the type or class of the vessel. Even for the same surface craft, the noise spectra vary widely depending on the length of the noise data waveform. The vessel-generated noises of intermittent nature, coupled with the uneven vibrations of distorted shafts as well as the wear and tear of the moving mechanical structures due to aging effects, may lead to substantial variations in the spectral profiles computed using short-term and long-term noise data waveforms. The spectrally derivable features of a class of vessel gathered from long-term data may remain the same within the class, while the time-series analysis using short-term data may yield more source specific features. Thus, the spectra computed using a time-series autoregressive estimator, capable of supplying variable length data segments, is seen to have a strong dependence on the length of the noise data segment. This paper presents an approach for generating and updating a systematic source data record comprising all the spectrally decomposable source specific features, using variable length noise data segments.
Wave field synthesis, adaptive wave field synthesis and ambisonics using decentralized transformed control: Potential applications to sound field reproduction and active noise control
Authors: Gauthier Philippe-Aubert , Berry Alain , Woszczyk Wieslaw
Published: October 2013

Sound field reproduction finds applications in listening to prerecorded music or in synthesizing virtual acoustics. The objective is to recreate a sound field in a listening environment. Wave field synthesis (WFS) is a known open-loop technology which assumes that the reproduction environment is anechoic. Classical WFS, therefore, does not perform well in a real reproduction space such as room. Previous work has suggested that it is physically possible to reproduce a progressive wave field in-room situation using active control approaches. In this paper, a formulation of adaptive wave field synthesis (AWFS) introduces practical possibilities for an adaptive sound field reproduction combining WFS and active control (with WFS departure penalization) with a limited number of error sensors. AWFS includes WFS and closed-loop ‘‘Ambisonics’’ as limiting cases. This leads to the modification of the multichannel filtered-reference least-mean-square (FXLMS) and the filtered-error LMS (FELMS) adaptive algorithms for AWFS. Decentralization of AWFS for sound field reproduction is introduced on the basis of sources’ and sensors’ radiation modes. Such decoupling may lead to decentralized control of source strength distributions and may reduce computational burden of the FXLMS and the FELMS algorithms used for AWFS. [Work funded by NSERC, NATEQ, Université de Sherbrooke and VRQ.] Ultrasound/Bioresponse to
Broadband seismo-acoustic modeling in range-dependent layered media using the parabolic equation method
Authors: Calvo David C. , Collins Michael D.
Published: October 2013

We present an efficient broadband modeling capability based on synthesizing solutions of the frequency-domain wave equation computed using the wide-angle parabolic equation (PE) method. The PE [a version of NRL’s Range-dependent Acoustic Model (RAM)] takes into account recent advances for handling range-varying layer thicknesses, compression and shear waves, and surface/interfacial waves (Rayleigh, Stoneley). The technique allows for multiple sources on land, underground, or offshore. Long-range propagation examples demonstrating the computation of multiply reflected head waves are presented. Examples of using head waves and their related phenomena to perform geoacoustic inversion in multiple layers are shown.
Observations of zeroth-order head waves in the Yellow Sea
Authors: Dahl Peter H. , Choi Jee Woong
Published: October 2013

As part of the 1996 joint U.S.-China Experiment in the Yellow Sea, head wave arrivals were measured in waters 75 m deep and at range ∼700 m, on a vertical line array (VLA) that spanned the water column. Head waves are strongly linked with properties of the seabed, and thus provide a useful measure for geoacoustic inversion carried out at relatively short range. Based on a time-domain and beam-forming analysis of the VLA data, the sediment sound speed and its gradient are estimated, and the conclusion is made that the observed head waves are kinematically consistent with zeroth-order in their ray-series classification. This conclusion is verified by simulation using the RAM parabolic equation algorithm, and it is found that the dynamic properties of the data also point to a zeroth-order, rather than a first-order, classification. The classification in turn establishes a baseline model for the head wave energy spectrum that is compared with the measured data. It is found that the estimated energy spectrum agrees well with a zero-order model, i.e., proportional to the source spectrum, S(f), modified by sediment attenuation, rather than a first-order model, S(f)/f. [Research supported by ONR Ocean Acoustics.]
Determining the effects of absorption and aperture size on sound decay in a complex coupled volume system
Authors: Bradley David T. , Wang Lily M.
Published: October 2013

Coupled volume systems may consist of a large, dry main volume connected to a smaller, reverberant coupled volume through an acoustically transparent opening, known as a coupling aperture. Given the right combination of architectural parameters, a particular sound decay may be obtained in the main volume which exhibits the double-slope effect (DSE). DSE is characterized by a double-slope decay profile with a steep early decay and a shallow reverberant tail. This project focuses on two architectural parameters—absorption ratio between the main and coupled volumes, and aperture size—and how these parameters affect sound decay in a complex coupled volume system. A medium detail level computer model of a generalized imaginary concert hall has been constructed, and the computer modeling program ODEON has been used to predict the room impulse responses for several configurations of the architectural parameters in the model. Binaural auralizations have also been created in ODEON and used in subjective testing to determine the effects of parameter changes on subjective cues. Objective results, based on several DSE quantifiers, will be presented. Corresponding subjective results will also be discussed.
Development of a low profile acoustical door for use on racks and cabinets for the information technology industry
Authors: O’Connell Michael D. , Anderl William James
Published: October 2013

This paper presents the design of 19 inch rack acoustical doors balancing acoustical attenuation, airflow impedance and distribution in a short depth by combining air foil technology with acoustic baffle design. Design optimization was done utilizing fluid flow analytical modeling and verified with a air flow bench and an acoustical rack door test fixture. Higher heat loads in rack mounted computer equipment drive higher cooling requirements. In order to provide air cooling solutions, higher volumetric air flow is required resulting in higher acoustical noise levels. These noise levels can result in noise levels that are unacceptable to the customer. Acoustical doors lower noise levels but are prone to high flow impedance, uneven flow distribution and large physical depth. High impedances require higher air moving device speeds to offset the lost volumetric air flow. This decreases the effective acoustical attenuation. Various rack modules have different inlet and outlet air flow locations making the distribution of the air from the door (front) or into the door(rear) important. Solutions to these problems usually require large depths in order to provide blockage of line of site and gradual air flow lines to keep impedance low and provide even distribution of the air.
Simulating the acoustics of classical sound recording techniques in rooms
Authors: Braasch Jonas , Martens William L. , Marui Atsushi
Published: October 2013

Recently, a method to position sound sources in 3-D space using virtual microphone control has been proposed [J. Acoust. Soc. Am. 117, 2391]. In this computer-generated environment, gains and delays between a virtual sound source and virtual microphones are calculated according to their distances and the axis orientations of the microphone directivity patterns. In the follow-up study reported here, it was investigated how to best simulate the influence of a room on the virtual microphone recording. For this purpose, a virtual rectangular room was created using the mirror-image technique (up to second-order reflections). The room dimensions were copied from an existing concert space at McGill University. Late reverb was created using a multiple feedback delay network with a time-variant architecture to enable modulation. To evaluate the system, measured impulse responses between a sound source and a five-channel microphone setup were compared to the virtual impulse responses for the same room/microphone-placement configuration. Among the tested parameters was the signal ratio between direct sound source and reflections as a function of the microphone placement and choice of microphone directivity patterns. The coherence between the simulated microphone signals was adjusted as well from the measured data. [Work supported by NSERC and VRQ.]
Acoustics of a planetarium
Authors: Shepherd Micah , Leishman Timothy W. , Utami Sentagi
Published: October 2013

Brigham Young University has recently constructed a planetarium with a 38-ft.-diameter dome. The facility also serves as a classroom. Since planetariums typically have poor acoustics due to their domed ceiling structures, acoustical recommendations were requested before its construction. The recommendations were made in an attempt to create an acceptable listening environment for lectures and other listening events. They were based in part on computer models and auralizations intended to predict the effectiveness of several acoustical treatments on the outer walls and on the dome itself. The recommendations were accepted and the planetarium was completed accordingly. A series of acoustical measurements was subsequently made in the room and the resulting acoustical parameters were mapped over the floor plan. This paper discusses these results and compares them with the predictions of the computer models.
Arrival time analysis for seismic airgun data classification
Authors: Stephens James
Published: October 2013

Survey data were collected in order to characterize the output of a marine seismic airgun array for various propagation geometries. Near uniform sound speed in the survey area allowed for classification of first-break and secondary arrivals as direct arrival and either reflected or refracted path arrival, respectively, using simple propagation geometries. Plots of initial arrival versus horizontal offset between the airgun and receiver allow classification of the arrival as via direct, reflected, or refracted path in a manner similar to that employed in the analysis of seismic reflection data in the shot domain. Beyond a certain horizontal range (approximately 4.5 km), the critically refracted arrival overtakes the reflected arrival, affording an estimate of the propagation speed in the seafloor layer. Analysis of this sort allows separation of the direct arrival for analysis purposes.
Environmentally associated signal-to-interference variability in low-frequency active sonar in Korea Strait
Authors: Cable P. G. , Carey W. M.
Published: October 2013

An analysis has been conducted of spatial and temporal variability of target echo-to-interference measurements made during Area Characterization Test III (ACT III) in 1995 in the Korea Strait. The measurements were made over a 5-day period for five fixed bistatic geometries (spatial scale of order 200 square km) using explosive sources, bottom-mounted horizontal receiving hydrophone arrays, and passive reflector targets in 100-m-depth water under downward-refracting acoustic conditions. The bottom at the ACT III site was nearly flat, of sand–silt composition. Meteorological conditions were relatively calm over the conduction of the measurements, but the main proximate oceanographic feature, the South Korean Coastal Front, did result in a relatively varied and complex sound-speed environment. The signal-to-interference fluctuation statistics associated with a single ping was determined to give a standard deviation of order 1 dB, while the observed variability of signal-to-interference over the spatial and temporal scales of the measurements gave a standard deviation of about 2 dB. The observed variability of received signal-to-interference will be discussed in terms of physical causes and measurement error. [Work supported by ONR Code OA321.]
A review of spatial coherence and array signal gain results
Authors: Carey William M.
Published: October 2013

A question discussed over the last three decades is: ‘‘how does scattering in the deep and shallow-water-sound channels limit coherent array processing?’’ In particular, for the horizontal array, what is the limitation of resolution in terms of the mutual coherence function and its coherence length? For frequencies (400 Hz) the measurement of the magnitude-squared coherence in a multipath environment with a partly coherent-dynamic noise field is difficult. C. Carter has determined that the measurement of phase coherence in the presence of noise is a poor statistical estimator since it is a function of the signal-to-noise ratio with large confidence bounds [IEEE Trans. Audio Electroacoust. AU21, 388–389 (1973)]. However, array signal gain and a theoretical coherence functional form can also provide a measure of the horizontal coherence length. This paper reviews single path coherence results and those derived from array measurements and shows that representative deep-water lengths for a frequency of 400 Hz are 100 wavelengths at 400 km with a frequency scaling of the 5/2 power while for shallow water they are 30 wavelengths at 40 km. These numbers will be discussed in terms of internal wave scattering theories.
A hydro-acoustic source model in calculating noise field generated by breaking waves
Authors: Chen Xuemei , Means Steven L. , Szymczak William G. , Rogers Joel C. W.
Published: October 2013

To develop a complete model for the breaking wave noise, it is necessary to relate the source quantities to the physical parameters of the wave-breaking and noise-generation processes. In this paper, the source structure of an individual breaking wave is simulated using a coupled hydro-acoustic model, which incorporates the physical processes underlying the mechanisms of the generation of the noise. The physical processes of wave formation and breaking are modeled using a generalized hydrodynamics formulation, providing the hydrodynamic parameters, such as pressure variations and air cavity shapes, etc., for the acoustic calculation. In the acoustic simulation, an algorithm has been developed in handling wave propagation in irregular regions, such as the bubbly liquid generated by wave-breaking. For the noise field modeling, the locations and occurrence times for the individual breaking waves are specified as stochastic quantities using Poisson simulations and the total noise field is calculated as the superposition of the noise contributed by those breaking waves. [Work supported by ONR.]
Sound field temporal fluctuations due to mode coupling in shallow water with internal soliton
Authors: Badiey Mohsen , Grigorev Valery , Katsnelson Boris , Lynch James
Published: October 2013

Sound field temporal fluctuations due to mode coupling, resulting from traveling internal solitons (IS) approximately along the acoustic track is addressed. The spectrum of received signal amplitude fluctuations in the range of very low frequency (∼ F = 0.001–0.01 Hz) is obtained. Maximums in this spectrum, correspond to the times which IS passes the sound field ray cycles. A relationship between F-spectrum and acoustic frequency is illustrated by showing Frequency-frequency (F−f) diagram. It is shown that depending on the sound field frequency (f), different mode pairs give the most pronounced contribution to the sound intensity fluctuations. At the same time invariant frequency, F, of maximal fluctuations remains approximately the same for all sound frequency (f) bands. For example in the case of the SWARM95 experiment, F is about 0.0015 Hz for sound frequency, f in the range of 30–150 Hz. Generally, F depends on waveguide parameters and velocity of IS. In this paper, the physical meaning of the invariant frequency and its relationship to the sound frequency is discussed. In addition, the experimental setup for observation of F in future shallow-water acoustic experiments is proposed. [Work supported by CRDF, PFBR and ONR-321OA.]
Environmental influence on acoustic impulse responses across the New Jersey Shelf in the winter
Authors: Mignerey Peter C.
Published: October 2013

In December 2003 as part of the RAGS03 experiment, the Naval Research Laboratory moored three vertical arrays at ranges of 10, 20, and 30 km distant from fixed 300- and 500-Hz LFM acoustic sources. Data were recorded continuously for approximately 20 days, which covered periods of severe storms and various phases of internal tides. Associated temperature profile data show the internal tide arriving as either an internal jump or series of elevation waves superimposed on a bottom layer associated with the foot of a shelf-break front. Impulse responses derived from acoustic data received by the vertical arrays provide insight into the acoustic modal structure within the ocean for various sea states and phases of the internal tide. Results to date show significant loss of high modes during those time periods when either the internal tide is present or the sea state is high in comparison with periods of low sea state and absent internal tide. The extent of mode stripping increases with down-slope range. Apparently high acoustic modes are being scattered and stripped by near-boundary interactions. Evidence for these interactions will be presented along with a discussion of stripping processes. [Work supported by ONR.]
Acoustic sampling volume of a bistatic sonar
Authors: Foote Kenneth G.
Published: October 2013

The expression for the acoustic sampling volume of a monostatic sonar [K. G. Foote, J. Acoust. Soc. Am. 90, 959 (1991)] is generalized to the case of a bistatic sonar, with separate, non-collocated transmitting and receiving transducers. The bistatic sampling volume is the integral of a counting function over the physically accessible space. The argument of the counting function, a Heaviside step function, is the difference between the scattering amplitude, or other measure of scattering strength, and a noise-dependent threshold. When this difference is positive, the integrand is unity; when zero, one-half; when negative, zero. Both the sampling volume and a related quantity, the equivalent beam angle, are evaluated numerically for several bistatic geometries, transducer shapes, and narrowband frequencies for the cases of point scatterers and directional scatterers. The application of the bistatic acoustic sampling volume to the quantification of suspended particulate matter and aquatic organisms is discussed.
Generation of ultrasound radiation force with the use of time reversal acoustics principles
Authors: Sarvazyan Armen , Sutin Alexander
Published: October 2013

There are numerous medical applications of ultrasound radiation force (RF) which could be made more effective using the time reversal acoustics (TRA) principles. This paper gives an overview of research into physical and technical bases of RF generation in heterogeneous biological media using TRA focusing systems. A custom-designed compact multichannel TRA system for receiving, digitizing, storing, time reversing, and transmitting acoustic signals in a wide frequency range from 0.01 to 10 MHz has been developed and extensively tested in model systems and ex vivo tissues and bones. Shear strain and shear waves remotely induced in soft tissues and bones by radiation force were detected using various acoustical and optical means. Experimental studies fully confirmed the feasibility of TRA generation of RF and demonstrated several advantages over conventional means of remotely inducing shear stress in biological media. These advantages include a possibility to create highly localized (close to diffraction limit) shear stress in heterogeneous media stir focused ultrasound beam in 3-D volume using very simple hardware. [Work supported by NIH grant.]
Modulated acoustic radiation pressure and stress-coupling projections
Authors: Marston Philip L. , Thiessen David B.
Published: October 2013

Low-frequency deformation can be induced at a single frequency using radiation stress oscillations of double-sideband suppressed-carrier ultrasound [P. L. Marston and R. E. Apfel, J. Acoust. Soc. Am. 67, 27 (1980)]. The transducer voltage is proportional to a product of low- and high-frequency sine waves. To anticipate the shape and magnitude of induced deformations, it is helpful to expand the distribution of the radiation stress on the object to be deformed as a series of projections [P. L. Marston, J. Acoust. Soc. Am. 67, 15 (1980)]. Stress projections are also useful for unmodulated waves: the radiation force is an example. In addition to spherical and nearly spherical objects, recent experiments and calculations have concerned cylindrical objects [S. F. Morse, D. B. Thiessen, and P. L. Marston, Phys. Fluids 8, 3 (1996); W. Wei, D. B. Thiessen, and P. L. Marston, J. Acoust. Soc. Am. 116, 202 (2004)]. In standing waves the following projections are nonvanishing in the low acoustic frequency limit for appropriately situated dense objects: radial projection [M. J. Marr-Lyon, D. B. Thiessen, and P. L. Marston, Phys. Rev. Lett. 86, 2293 (2001)] and quadrupole projection [P. L. Marston et al., J. Acoust. Soc. Am. 69, 1499 (1981)].
The effects of choir spacing and choir formation on the tuning accuracy and intonation tendencies of a mixed choir
Authors: Daugherty James F.
Published: October 2013

The tuning accuracy and intonation tendencies of a high school mixed choir (N=46) were measured from digital recordings obtained as the ensemble performed an a cappella motet under concert conditions in N=3 singer spacing configurations (close, lateral, circumambient) and N=2 choir formations (sectional and mixed). Methods of analysis were modeled on Howard’s (2004) pitch-based measurements of the tuning accuracy of crowds of football fans. Results were discussed in terms of (a) previous studies on choir spacing (Daugherty, 1999, 2003) and self-to-other singer ratios (Ternstrm, 1995, 1999); (b) contributions of choir spacing to vocal/choral pedagogy; and (c) potential ramifications for the design and use of auditoria and portable standing risers for choral performances.
Auditorium design for choral performance
Authors: Foulkes Timothy , Storch Christopher
Published: October 2013

Design for a 500 seat recital hall to support an award winning high school choral program is discussed. Acoustic design strategy, important acoustic parameters (calculated and measured), and photos of the completed project are reviewed.
Peter Ladefoged and phonetics in the field
Authors: Maddieson Ian
Published: October 2013

Among many distinctive contributions to phonetics by Peter Ladefoged is an insistence on the immense diversity of phonetic phenomena in the languages of the world, particularly at the segmental level. Because of this Peter has maintained a flexible approach to any scheme of classification or description, adapting to both new approaches and new data. Perhaps more than any other phonetician he has expected to find surprises, and has gone to far corners of the world in search of them. His ground-breaking Phonetic Study of West African Languages from 1964 laid out a template for synthesizing a large mass of data which is echoed in later works such as Preliminaries to Linguistic Phonetics and Sounds of the World’s Languages. His widely used Course in Phonetics and other textbooks have shown generations of students the richness of spoken sound. But not only novel research results have marked Peter’s contributions to this aspect of phonetics; equally significant are his methodological innovations in taking experimental techniques to the field. Some of the most salient steps along both these paths will be reviewed from personal and professional perspectives.
Pronunciation models for conversational speech
Authors: Johnson Keith
Published: October 2013

Using a pronunciation dictionary of clear speech citation forms a segment deletion rate of nearly 12% is found in a corpus of conversational speech. The number of apparent segment deletions can be reduced by constructing a pronunciation dictionary that records one or more of the actual pronunciations found in conversational speech; however, the resulting empirical pronunciation dictionary often fails to include the citation pronunciation form. Issues involved in selecting pronunciations for a dictionary for linguistic, psycholinguistic, and ASR research will be discussed. One conclusion is that Ladefoged may have been the wiser for avoiding the business of producing pronunciation dictionaries. [Supported by NIDCD Grant No. R01 DC04330-03.]
Introduction to deconvolution
Authors: Jansson Peter A.
Published: October 2013

Deconvolution tasks will always lie at the frontier of human knowledge in many fields, almost by definition. Rising in the latter 20th century from near disreputability, first to usefulness, then necessity in some disciplines, the varied techniques of deconvolution have assumed an important role in the scientists tool kit. This talk will trace deconvolutions development with examples, including many ‘‘firsts,’’ drawn from spectroscopy, radio astronomy, photography, cell biology, color science and diverse other fields. Following a tutorial introduction, detail will be provided on modern super-resolving methods and lesser known topics such as selected-ordinate image (SORI) processing.
From Philharmonic Hall to number theory: The way to more diffusion
Authors: Schroeder Manfred R.
Published: October 2013

In September 1962, in the presence of Mrs. Jacqueline Kennedy, Philharmonic Hall in New York was inaugurated—the first building of the new Lincoln Center for the Performing Arts. To address the soon-apparent acoustic problems, Lincoln Center turned to Bell Laboratories for help, and I was asked to join a ‘‘committee of experts,’’ chaired by Vern O. Knudsen of UCLA. My work on Philharmonic Hall, assisted by B.S. Atal, G.M. Sessler, and J.E. West, and later, after my move to Göttingen, by my students D. Gottlob, F.K. Siebrasse, and U. Eysholdt, indicated a need for energetic early lateral sound. It was clear that better lateral diffusion could improve the acoustic quality and the feeling of ‘‘envelopment’’ by the sound. Knowing some Galois field mathematics, I lucked upon the design of diffusors which scattered incident waves into broad lateral patterns—but only for a single musical octave. Then, in 1977, during a celebration of the 200th anniversary of Gauss’s birth, I heard a talk by André Weil on Gauss sums and quadratic residues and, in a flash, it became clear to me that diffusors based on quadratic residues were the answer to broadly scattering waves comprising many musical octaves.
A largely diffuse small room
Authors: Massenburg George , D’Antonio Peter
Published: October 2013

An 8 m×10 m rectangular room is described that will have broad bandwidth diffusion completely covering 5 of its 6 surfaces, i.e., all walls and ceiling surfaces. The dimensions of the room were optimized for minimal standard deviation of the modal response. The wall surfaces are treated with a large prime, single period, number theoretic diffusor 1.2 m deep that wraps around the entire room. The ceiling consists of a 12×13 low frequency diffusor 2 m deep, which is further treated with mid-high frequency diffusors to form a nested, diffractal surface. These surfaces will be described further and illustrated. The room is intended to be deployed as a monitor room for mixing surround sound. It is hoped that the unique combination of a reduced number of specular surfaces and very neutral, wide-band ambience will improve localization (particularly for virtual sources) and offer greater support for (and speed up) the balancing/mixing of multitrack sources. The perceived effectiveness of the diffusor in the ceiling, particularly in the 2 low octaves will be described. The room will also be used as a massively diffuse recording room for various musical formats.
Making an anechoic choral recording
Authors: Freiheit Ron , Alexander John , Ferguson John
Published: October 2013

The utilization of auralization as a tool for acoustic analysis continues to grow and develop. An important element for successful auralization listening experiences is the selection of anechoic source material. In researching the current library of anechoically recorded source material, it was discovered that choral material was not readily available. The Wenger Corporation, St. Olaf College, and 3M undertook a joint project to create an anechoic choral recording. The paper describes the challenges of this recording project—from the technological, logistical, and musical standpoints—and the solutions that were successfully implemented.
Nonlinear acoustic behaviors of soils
Authors: Lu Zhiqu
Published: October 2013

Soil, like other porous materials such as rock, sand, sandstone, sedimentation, and concrete, exhibits remarkable nonlinear behaviors. In this work, the nonlinear acoustic behaviors of soils are studied by measuring both amplitude and phase frequency responses as a function of dynamic strain. Typical nonlinear phenomena such as resonant frequency shift and harmonics generation are observed. It is also found that the phase difference between the received and reference signals changes with strain. New methods for measuring the nonlinearity parameters, based on resonant frequency shift, phase shift, and phase slope measurements, are developed and compared. This suggests that phase could be a very useful parameter for determining the nonlinearity of porous materials. [This work is supported by the USDA Agricultural Research Service under Specific Cooperative Agreement 58-6408-0-108.]
Nonlinear acoustic experiments involving landmine detection: Connections with mesoscopic elasticity and slow dynamics in geomaterials, Part III
Authors: Korman Murray S. , Sabatier James M.
Published: October 2013

In nonlinear acoustic detection schemes, airborne sound at two primary tones, f1, f2 (closely spaced near an 80-Hz resonance) excites the soil surface over a buried landmine. Due to soil wave interactions with the landmine, a scattered surface profile can be measured by a geophone. Profiles at f1, f2, f1−(f2−f1) and f2+(f2−f1) exhibit single peaks; those at 2f1−(f2−f1), f1+f2 and 2f2+(f2−f1) involve higher order mode shapes for a VS 2.2 plastic, inert, anti-tank landmine, buried at 3.6 cm in sifted loess soil [J. Acoust. Soc. Am. 116, 3354–3369 (2004)]. Near resonance, the bending (softening) of a family of increasing amplitude tuning curves, involving the vibration over the landmine, exhibits a linear relationship between the peak particle velocity and corresponding frequency. Results are similar to nonlinear mesoscopic/nanoscale effects that are observed in granular solids like Berea sandstone. New experiments show that first sweeping up through resonance and then immediately sweeping back down result in different tuning curve behavior that might be explained by ‘‘slow dynamics’’ where an effective modulus reduction persists following periods of high strain. Results are similar to those described by TenCate et al. [Phys. Rev. Lett. 85, 1020–1023 (2000)]. [Work supported by U.S. Army RDECOM CERDEC, NVESD.]
Nonlinear acoustic landmine detection: Comparison of ‘‘off target’’ soil background and ‘‘on target’’ soil-mine nonlinear effects
Authors: Korman Murray S.
Published: October 2013

When airborne sound at two primary tones, f1, f2 (closely spaced near a resonance) excites the soil surface over a buried landmine, soil wave motion interacts with the landmine generating a scattered surface profile which can be measured over the ‘‘target.’’ Profiles at f1, f2, and f1−(f2−f1), f2+(f2−f1), 2f1−(f2−f1), f1+f2 and 2f2+(f2−f1) (among others) are measured for a VS 1.6 plastic, inert, anti-tank landmine, buried at 3.6 cm in sifted loess soil. It is observed that the ‘‘on target’’ to ‘‘off target’’ contrast ratio for the sum frequency component can be ∼20 dB higher than for either primary. The vibration interaction between the top-plate interface of a buried plastic landmine and the soil above it appears to exhibit many characteristics of the mesoscopic/nanoscale nonlinear effects that are observed in geomaterials like sandstone. Near resonance, the bending (softening) of a family of increasing amplitude tuning curves, involving the vibration over the landmine, exhibits a linear relationship between the peak particle velocity and corresponding frequency. Tuning curve experiments along with two-tone tests are performed both on and off the mine in an effort to understand the nonlinearities in each case. [Work supported by U.S. Army RDECOM CERDEC, NVESD.]
A study of the acoustic behavior of a plastic, blast-hardened, antitank landmine
Authors: Alberts W. C. Kirkpatrick , Waxler Roger , Sabatier James M.
Published: October 2013

Mechanical resonances in landmines are often exploited by some acoustic-to-seismic detection schemes. As such, a study of an acoustically active landmine has been performed. Measurements of the mechanical resonances of the pressure plate reveal seven modes below 1.6 kHz and a large frequency shift of the first symmetric mode of the pressure plate, such that it occurs higher in frequency than the first asymmetric mode. The pressure plate is considered to act as an elastically supported thin plate, and the observed perturbation of the first symmetric mode is believed to be caused by the landmine’s blast-hardening mechanism. This is confirmed by a lumped acoustic element model of the system of pressure plate and blast-hardening mechanism. Mass-loading experiments in water show a decrease in resonance frequency with increasing depth that saturates at shallow depths as predicted by Velea et al. [J. Acoust. Soc. Am. 115, 1993 (2004)]. Similar mass-loading measurements in sand reveal an interesting phenomenon; flush burial of the landmine effectively removes the effect of the blast-hardening mechanism on the pressure plate. A qualitative theoretical model of this phenomenon will be presented.
Muscular hydrostat mechanism for lip protrusion in speech
Authors: Murano Emi Z. , Stone Maureen , Honda Kiyoshi
Published: October 2013

The lip is an organ consisting mostly of muscle similar to the tongue. While the tongue is known as a muscular hydrostat, it is unclear whether the lip is also. In this paper the muscular hydrostat issue was explored from the anatomical and functional point of view using high-resolution static MRI (hr-MRI; 0.125 mm/pixel) and tagged-cineMRI (t-MRI). A 3-D reconstruction of the lips and its muscles was obtained from hr-MRI during sustained vowels /i/ and /u/. The muscular geometry of the orbicularis oris, mentalis, and depressor labii inferior muscles were superimposed onto the principal strains that depicts compression and expansion of the internal tissue obtained from t-MRI. It is shown that (1) orbicularis oris muscle shape can predict both the borderline of glabrous and hairy skin and the manner in which the lips are protruded; (2) the lips volume is almost identical for both speech tasks; and (3) direction and intensity of compression of orbicularis oris and mentalis muscle bundles imply the role of these muscles in the protrusion appearance. These results indicate that the muscular architecture and volume preserving characteristics of the lips are consistent with a muscular hydrostat. [This work was supported by NIH (USA) and NiCT (Japan).]
Tongue movements in vowel-consonant-vowel (VCV) sequences: The effect of consonant length
Authors: Lofqvist Anders
Published: October 2013

This study examined the effect of consonant duration on the tongue movement from the first to the second vowel in VCV sequences, where the consonant is a short or long labial nasal consonant. Lip, tongue, and jaw movements were recorded in native speakers of Japanese using a magnetometer system. Measurements were made of the duration, path, and speed of the tongue movement trajectory between the two vowels. The coordination of the onsets of the lip closing and tongue movements was also studied, as well as the relative part of the trajectory that occurred during the consonant and the vowels. Preliminary results show a robust difference in duration between the long and short consonants, with the long ones about twice as long. The duration of the tongue movement was longer in the long than the short consonants. Both the peak and average speed of the tongue movement were slower in the long consonants. The tongue movement path was slightly longer in the long consonants. These results suggest that speakers adjust the tongue movement trajectory so that a similar relationship between the movement and the consonant closure is maintained in both the long and the short consonants. [Work supported by NIH.]
Simulating and understanding the effects of velar coupling area on nasalized vowel spectra
Authors: Pruthi Tarun , Espy-Wilson Carol Y.
Published: October 2013

MRI-based area functions for the nasal cavity of one speaker were combined with the area functions for the vowels /iy/ and /aa/ to study nasalized vowels. The oral cavity was compensated for the falling velum by decreasing the oral cavity area by an amount equal to the increase in the nasal cavity area. Susceptance plots were used along with the simulated transfer functions to understand the effects of velar coupling on nasalized vowel spectra. Susceptance plots of −(Bp+Bo) and Bn suggested significant deviation from the rules suggested by O. Fujimura and J. Lindqvist [J. Acoust. Soc. Am. 49(2), 541–558 (1971)]. In particular, the plots showed that: (1) the frequency of zero crossings of the susceptance plots changes with a change in the coupling area, and (2) formant frequencies need not shift monotonically upward with an increase in coupling area. Further, as a consequence of (1), and the fact that an increase in the coupling area results in a shift of Bn to the right and −(Bp+Bo) to the left, it is postulated that zero crossings of the two plots can cross each other. [MRI data from Brad Story. Work supported by NSF Grant No. BCS0236707.]
Recognition of voice onset time for use in pronunciation modeling
Authors: Kazemzadeh Abe , Lee Sungbok , Narayanan Shrikanth
Published: October 2013

This study examines methods for recognizing native and accented voiceless stops based on voice onset time (VOT). These methods are tested on data from the Tball corpus of early elementary school children, which includes both native English speakers and Spanish speakers learning English, and which is transcribed to highlight pronunciation variation. We examine the English voiceless stop series, which have long VOT and aspiration, and the corresponding voiceless stops in Spanish accented English, which have short VOT and little aspiration. The methods tested are : (1) to train hidden Markov models (HMMs) based on native speech and then extract the VOT times by post-processing phone-level alignments, (2) to train HMMs with explicit aspiration models, and (3) to train, for each phoneme, different HMMs for native and accented variants. Error rates of 23%–53% for distinguishing phone VOT characteristics are reported for the first method, 5%–57% for the second method, and 0%–36% for the third. The error rates varied depending on the different phones examined. In general, the /p/ and /k/ phones had results that varied more than /t/. These results are discussed in light of each method’s usefulness and ease of implementation, and possible improvements are proposed.
Improving automatic speech recognition via better analysis and adaptation
Authors: O’Shaughnessy Douglas , Wang Wayne , Zhu William , Barreaud Vincent , Nagarajan T. , Muralishankar R.
Published: October 2013

One way to improve automatic speech recognition (ASR) systems is to reduce the mismatch between system training and operating conditions, as such mismatch seriously degrades performance. We have developed model adaptation techniques able to adapt to various speech environments without modifying ASR systems, and have developed an appropriate feature transformation scheme for the Mel-frequency cepstral coefficients (MFCC), a popular front-end feature of ASR systems. We use maximum a posteriori model adaptation and a method based on Bayesian parametric representation. Feature transformation aims to maximize the desired source of information for a given speech signal in the front-end features and to minimize undesired sources. Frequency-domain autoregressive modeling and a segmentation algorithm are being developed, e.g., to segment a speech signal into syllablelike units. We also introduce a new speech-processing front-end feature that performs better than the existing MFCC, as well as a log-energy dynamic range normalization technique for ASR in adverse conditions. In addition, we have developed a continuous ASR method that exploits the advantages of syllable and phoneme-based subword unit models. [Work supported by NSERC-Canada and Prompt-Quebec.]