bluuebunny/crossref_metadata_2025 huggingface dataset:
The effects of prior access to talker information on vowel identification in single- and mixed-talker contexts
Authors: Morton John R. , Lulich Steven M. , Sommers Mitchell
Published: September 2012

Speech intelligibility is significantly impaired when words are spoken by multiple- compared with single talkers. In the present study, we examined whether providing listeners with information about the vocal tract characteristics of the upcoming speaker would reduce the difference between single- and mixed-talker conditions. All participants were initially trained to identify 6 talkers (3 male and 3 female) from isolated vowels. Participants then completed closed-set vowel identification tests, including a blocked and a mixed condition, and one of two mixed-talker precursor conditions. In one precursor condition, participants saw one of the six talker’s name immediately prior to hearing the target vowel. In the other precursor condition, participants heard a non-target vowel (/i/) spoken immediately before the target stimulus. For both precursor conditions, the name (or vowel) precursor matched the target-vowel speaker on half of the trials. For the other half, the precursor was of the same gender as the target-vowel speaker. Only when a sample vowel precursor was spoken by the same talker as the subsequent target was there a significant improvement in scores relative to the mixed-talker condition. The results suggest that exposure to isolated vowels can provide enough information about that talker’s vocal tract to improve perceptual normalization.
The role of first-language production accuracy and talker-listener alignment in second-language speech Intelligibility
Authors: Lee Kyounghee , Bradlow Ann R.
Published: September 2012

This study investigated variability in L2 speech intelligibility as a function of L1 speech intelligibility and of talker-listener L1 match. Non-native Korean talkers varying in their L2 proficiency were recorded reading simple English (L2) and Korean (L1) sentences. The intelligibility of these sentences was then assessed by Korean listeners (both Korean and English productions) and English listeners (English productions only) in a sentence recognition task. The results revealed that for these Korean-English bilingual talkers, L1 intelligibility was significantly correlated with L2 intelligibility for both Korean and English listeners, suggesting that variability in L1 speech intelligibility can serve as a predictor of variability in L2 production accuracy. We also examined the interlanguage speech intelligibility benefit for non-native listeners (ISIB-L) (e.g., Bent & Bradlow, 2003; Hayes-Harb et al., 2008). Korean listeners performed better at identifying the sentences produced by Korean talkers with relatively low L2 intelligibility, implying that the benefit of a shared native language between talkers and listeners may be larger when non-native listeners process speech from a talker with low L2 intelligibility. Overall, these findings indicate that variability in L2 speech intelligibility is related to language-general talker characteristics as well as to the talker-listener language alignment.
Speech recognition informed by distinctive feature theory: The featurally underspecified Lexicon model and its implications
Authors: Roberts Philip J. , Reetz Henning
Published: September 2012

We present a speech recognition engine that implements the Featurally Underspecified Lexicon calculus (FUL). The FUL model defines an inventory of privative phonological features that is necessary and sufficient to describe contrasts between phonemes in any language in the world. The model also defines conditions for comparing feature bundles recovered from the signal with segments defined in the lexicon: a feature may MATCH (the feature is present in both the signal and the lexicon), it may MISMATCH (the feature in the signal is impossible in tokens of the segment in the lexicon, e.g. when a stop-burst, which indicates the [PLOSIVE] feature, is compared with a segment carrying the [CONTINUANT] feature in the lexicon), or it may provoke a NOMISMATCH response when the feature in the signal is not part of the segment in the lexicon. This matching mechanism also accounts for asymmetries as they are observed in natural speech, as [CORONAL] assimilates to [LABIAL] but not the other way around. The engine computes distances to neighboring words according to a coherence measure to simulate co-activation in the lexicon. We will demonstrate online the operation of this engine in English and German.
Evaluating automatic speech-to-speech interpreting
Authors: Bernstein Jared , Rosenfeld Elizabeth
Published: September 2012

A naïve speech-to-speech interpreter can be implemented as three component processes in series: speech recognition, machine translation, and speech synthesis. However, evaluating the performance of a speech-to-speech interpreting system may be either simpler or more complicated than merely calculating the product of the accuracies of those three component processes. This is because human users are sensitive to the rate at which an interpreter operates and because a system’s communication success rate is properly measured by a listener’s correct understanding of the speaker’s intention in a particular context rather than by the system’s word-for-word accuracy and intelligibility. We are evaluating two currently available systems, each operating in both directions for two language pairs: English/Spanish and English/Chinese. For each system and each language pair and each direction, we compare word-for-word spoken interpretation accuracy with the successful communication of speaker intent-in-context. Results for the Spanish/English pair suggest that word-for-word accuracy is high (about 75-90% correct) for both systems, and that taking a lower information rate measure like communication of intent in context reduces the error rate substantially. Finally, suggestions for improved system design are presented.
Parametric forms of spectral amplitude nonlinearities for use in automatic speech recognition
Authors: Zahorian Stephen
Published: September 2012

This work is a continuation and extension of work presented at the fall 2011 meeting of the Acoustical Society of America (Wong and Zahorian). In that work, and also at work done at Carnegie Mellon University, auditory model derived spectral amplitude nonlinearities, with symmetric additional compression (after log amplitude scaling) were found to improve automatic speech recognition performance when training and test data are mismatched with respect to noise. In this new work, a parametric nonlinearity, controlled by three parameters, was formed to allow non-symmetric compression with respect to high and low amplitudes. Several variations of this basic nonlinearity were evaluated for both matched and mismatched conditions for training and test data with respect to noise. For the mismatched cases, the most effective nonlinearity was found to be compressive for low amplitudes but expansive at high amplitudes (thus emphasizing spectral peaks). However, for matched conditions, none of the spectral amplitude nonlinearities improve automatic recognition accuracy. The additional nonlinearity can be combined with log compression to create a single unified amplitude nonlinearity for speech processing.
Acoustical features in Mandarin emotional speech by native speakers of English
Authors: Jian Hua-Li , Hung Jessica
Published: September 2012

This study examines (1) whether native speakers of English (NS-E) can express emotions successfully in Mandarin speech, and (2) how their emotional expressions differ from native speakers of Mandarin (NS-C) when the emotional portrayals are recognizable. The acoustic features analyzed included F0, duration, and intensity. The scenario approach was adopted to elicit emotions joy, anger, sadness and fear, with neutral as a control. The data gathered (Sixteen NS-E and NS-C) were rated. F0 range at sentential level, mean F0 of each syllable, sentential and syllabic duration, and intensity signal of each segment were contrasted across groups within each emotional expression. The findings indicated that emotions by NS-C were recognized well, but joy, anger and fear by NS-E had low recognition rates because of accents, vocal cues and culture-specific components. Both groups adopted similar F0 range at the sentence level but joy in both and fear by NS-C showed small range. Only NS-C showed fast speech rate in anger. Emotions with high activations by NS-E were shorter. Anger and joy showed high intensity, while sadness and fear low intensity in both groups. NS-C tended to use different intensity range to indicate different emotions, while NS-E used similar range for all emotions.
Cues for the perception of expressive speech
Authors: Hubbard Daniel J. , Assmann Peter F.
Published: September 2012

In a previous study the contribution of fundamental frequency (F0) to the perception of expressive speech was examined using a selective adaptation technique. Listeners heard either F0-present or F0-removed adaptors (vocoder-processed VCV syllables) from one of two expressive categories: angry or happy. In a two-alternative listening task, contrastive aftereffects (characterized by a tendency to label test stimuli as originating from the non-adapted category) were documented only in the F0-present condition. This suggests that F0 is important for the perception of emotional expressions. However, listeners were still able to identify F0-removed syllables as angry or happy at a rate significantly better than chance (58%). Subsequent analyses revealed systematic differences in formant frequencies as potential cues for categorization (higher F1 and F2 frequencies for happy compared to angry tokens). A discriminant analysis was performed using formant measurements (F1-F3) and measures related to the voicing source. Removing voice source measures (including mean F0) produced a decrease in classification accuracy that closely matched listener response patterns for the F0-removed stimuli. The results suggest that in the absence of F0, formant frequencies may be used for perception of angry and happy speech.
Partial effects of perceptual compensation need not be auditorily driven
Authors: Finley Gregory
Published: September 2012

An experiment was devised to test whether compensation for coarticulation could be motivated by nonspeech for which gestural recovery is impossible. Subjects were presented with CV stimuli formed by concatenating an /s ~ ∫/ continuum fricative with an /i/ or /o/ from three types of synthesized vocalic nuclei: full spectral vowels (Set A), vowels with F2 but no other formants (B), and pure sine tones at F2 (C). F2 was chosen as the common parameter between sets because extremely high or low F2 is enough information for English speakers to judge vowel roundedness (and rounding lowers centroid frequency, a key difference between /s/ and /∫/). Comparing fricative identification boundaries between the vowels within each set, compensation occurred reliably in Sets A (t = 4.6, p < 0.01) and B (t = 3.7, p < 0.01) but not C. Comparing conditions against each other, Set A showed significantly higher vowel-triggered boundary shift than B (F = 8.18, p < 0.05). Results support the conclusion that relevant acoustic cues might not trigger compensation if they cannot be associated with speech; additionally, sounds reminiscent of speech do not generate as strong an effect as sounds firmly identifiable as speech.
Language specific compensation for coarticulation
Authors: Johnson Keith , Kang Shinae , Finley Greg , Miller Rigoli Carson , Spinelli Elsa
Published: September 2012

This paper reports an experiment testing whether compensation for coarticulation in speech perception is mediated by linguistic experience. The stimuli are a set of fricative-vowel syllables on continua from [s] to [∫] with the vowels [a], [u], and [y]. Responses from native speakers of English and French (20 in each group) were compared. Native speakers of French are familiar with the production of the rounded vowel [y] while this vowel was unfamiliar to the native English speakers. Both groups showed compensation for coarticulation (both t > 5, p < 0.01) with the vowel [u] (more “s” responses indicating that in the context of a round vowel, fricatives with a lower spectral center of gravity were labeled “s”). The French group also showed a compensation effect in the [y] environment (t[20] = 3.48, p<0.01). English listeners also showed a tendency for more subject-to-subject variation on the [y] boundary locations than did the French listeners (Levene's test of equality of variance, p < 0.1). The results thus indicate that compensation for coarticulation is a language specific effect, tied to the listener's experience with the conditioning phonetic environment.
Effects of visual cue enhancement on speech intelligibility for clear and conversational speech in noise
Authors: Beitz Jasmine , Van Engen Kristin , Smiljanic Rajka , Chandrasekaran Bharath
Published: September 2012

Visual presentation of a speaker enhances the auditory perception of speech information in noisy conditions (e.g. Helfer and Freyman, 2005; Helfer, 1997). Intelligibility is also improved when a speaker adopts a clear speaking style (Smiljanic and Bradlow, 2009). The present study investigates the contributions of these two intelligibility-enhancing factors in the presence of several types of noise, which vary with respect to the degree of informational and energetic masking they impose on target speech. Specifically, it measures sentence intelligibility in the presence of 1 competing talker, 2-talker babble, 4-talker babble, and speech-shaped noise. Meaningful sentences (in clear and conversational styles) were presented to participants in each modality (audio-only; audio-visual) and in all noise conditions. Participants reported all intelligible words. Our data shows overall better intelligibility for clear speech and for AV speech relative to conversational speech and audio-only condition. However, the visual benefit associated with conversational speech is significantly greater than the visual benefit associated with clear speaking style. The relative contribution of visual influences and speech clarity to intelligibility enhancements will be discussed.
Performance of a large-aperture array in a complex littoral environment
Authors: Means Steven L. , Wales Stephen C. , Rogers Jeffrey S.
Published: September 2012

Over 850 hours, from an ~2 month time period, of ambient acoustic measurements were taken on a long (917m), 500 phone linear array 12 km off the coast of Fort Lauderdale, Florida, capturing both day and night, commercial and recreational shipping generated noise. Marine-band radar and AIS data were collected concurrently so that ship locations could be tracked in a large area surrounding the array. Array performance is investigated by beamforming in a number of frequency bands and apertures to determine median beam noise, noise gain reduction, and noise window statistics as a function of bearing. Additionally, the mean-square coherence is computed as a function of normalized distance (range/wavelength). The results are compared for a variety of time periods and environmental conditions. Comparisons of measurements with a computational noise model using known ship locations will be made. (Work supported by ONR base funding at NRL.)
Imaging objects with variable offset in an evanescent wave field using circular synthetic aperture sonar and spectroscopy
Authors: Plotnick Daniel , Marston Philip L.
Published: September 2012

Imaging properties of objects suspended in an acoustic evanescent wave field are examined. Evanescent waves are generated using a tank containing immiscible liquids and an appropriately directed acoustic beam [C. F. Osterhoudt et al., IEEE J. Oceanic Eng. 33, 397-404 (2008)]. The source and receiver transducers are in the liquid having the higher sound velocity. Object(s) are spun about a vertical axis while scattering is measured. The object(s) offset into the wave field is then varied and the experiment repeated. In this work small spheres and other objects are used to gain insight into imaging properties as a function of the object or object(s) position in the evanescent field. Data is examined using circular synthetic aperture techniques. Additionally, a comparison is made between spectral data and a heuristic model in the case of two spheres. The spectral evolution in that case is affected by the interference from the two scatterers. Cases where the source and receiver are collocated (monostatic) and those where the source and receiver are separated (bistatic) are compared. [Work supported by ONR.]
Blind source localization and separation in three-dimensional space
Authors: Zhu Na , Wu Sean
Published: September 2012

A new methodology called blind source localization and separation (BSLS) is developed to locate sound sources in three-dimensional space and extract their corresponding sound signals from the directly measured data. This technology includes two steps: Firstly, locate the sound sources by applying signal pre-processing and triangulation algorithms to the signals measured at microphones. Secondly, taking the sound source location results from the first step and measured signals at microphone as the input data, use the point source separation method to extract the sources and reconstruct the sound sources at their locations. The impact of various factors, such as the types and characteristics of sources, microphone configurations, signal to noise ratios, number of microphones, and errors in source localizations on the quality of source separation will be examined and compared to those obtained by the conventional blind source separation method.
Broadband sparse-array blind deconvolution using unconventional beamforming
Authors: H. Abadi Shima , Song Heechun , R. Dowling David
Published: September 2012

Sound from a remote underwater source is commonly distorted by multipath propagation. Blind deconvolution is the task of estimating the unknown waveforms for the original source signal, and the source-to-receiver impulse response(s), from signal(s) recorded in an unknown acoustic environment. Synthetic time reversal (STR) is a technique for blind deconvolution of underwater receiving-array recordings that relies on generic features of the propagating modes or ray paths that lead to multipath sound propagation. In prior studies the pivotal ingredient for STR, an estimate of the source-signal's phase (as a function of frequency), was generated from conventional beamforming of the recorded signals. However, through the use of unconventional nonlinear frequency-difference beamforming, STR can be extended to sparse array recordings where the receiving-array elements are many wavelengths apart and conventional beamforming is inadequate. This extension of STR was tested with simple propagation simulations and FAF06 experimental measurements involving broadband signal pulses (11-19 kHz) that propagate 2.2 km in the shallow ocean to a vertical 16-element receiving array having a 3.75-m-spacing between elements (almost 40 signal-center-frequency wavelengths). The cross-correlation coefficient between the source-broadcast and STR-reconstructed-signal waveforms for the simulations and experiments are 98% and 91-92%, respectively. [Sponsored by ONR.]
Beam steering response matrix inversion method: Accurate localization of simultaneous high and low frequency sources using a small line array
Authors: Mooney Jon W.
Published: September 2012

The Beam Steering Response (BSR) Matrix Inversion Method is a technique using a small line array that, when mathematically steered to populate a BSR matrix, can simultaneously resolve the locations of multiple sources. Using this technique, high directivity over a wide frequency range may be achieved with a small array. Simultaneous tracking of multiple low and high frequency sources is demonstrated using the BSR Matrix Inversion Method with an acoustic video camera having a 49 inch, 8 element line array. This paper addresses 1) the derivation of the Beam Steering Response matrix, 2) line array design considerations, 3) processing limitations, and 4) a practical demonstration of the BSR Matrix Inversion Method.
An experimental study of drive-by single channel time reversal using scaled targets and clutter
Authors: Abawi Ahmad T. , Kirsteins Ivars , Plotnick Daniel
Published: September 2012

Previous results suggest that iterative single channel time reversal (TR) is a promising, simple and inexpensive technique to enhance the backscatter signature of elastic objects and to simultaneously focus on its dominant resonance frequency in the presence of noise and clutter (Pautet et al. 2005), (Waters et al. 2009). However, to the best of our knowledge, these and other studies have only considered the case when the sonar system, target, and clutter are fixed or stationary, i.e., not moving. This is an unrealistic assumption since the sonar platform is typically moving. In fact, motion and environmentally-induced clutter scintillations may actually be beneficial for TR when the dominant target resonance varies slowly with aspect in clutter-limited noise environments. Theoretical arguments in (Kirsteins et al. 2009) suggest TR provides additional signal-to-noise ratio (SNR) enhancements against clutter under these circumstances. To confirm this hypothesis, a drive-by TR experiment was conducted in a tank using scaled elastic targets/clutter sources with the platform motion simulated by rotating the target/clutter during each TR iteration. Quantitative analysis in regards to detecting the dominant resonance clearly shows that TR in drive-by mode provides SNR enhancements against clutter.
Estimation of motion parameters and trajectory of an low flying aircraft using acoustic sensors
Authors: Saravanakumar A.
Published: September 2012

An aircraft generates an acoustic impulse that propagates outwards from the source. The position of the source and hence the trajectory can be estimated by measuring the relative time of arrival of the impulse at a number of spatially distributed sensors. The time difference for the acoustic wave front to arrive at two spatially separated sensors is estimated by cross correlating the digitized outputs of the sensors. The time delay estimate is used to calculate the source bearing and the position of the source is found using triangulation technique using the bearings from two widely separated receiving nodes. The flight parameter of the aircraft is obtained by autocorrelation method using acoustic Multipath delays. The signal emitted by an UAV arrives at a stationary sensor located above a flat ground via a direct path and a ground-reflected path. The difference in the times of arrival of the direct path and ground-reflected path signal components is known as the Multipath delay. A model is developed to predict the variation with time and autocorrelation method is formulated to estimate the motion parameters like speed and altitude of the aircraft.
Accuracy of passive source localization approaches with a single towed horizontal line-array in an ocean waveguide
Authors: Gong Zheng , Ratilal Purnima
Published: September 2012

Instantaneous passive source localization applying the (1) synthetic aperture tracking, (2) array invariant, (3) bearings-only target motion analysis in modified polar coordinates via the extended Kalman filter, and (4) bearings-migration minimum mean-square error methods to measurements made on a single towed horizontal receiver array in a random range-dependent ocean waveguide are examined. These methods are employed to localize and track a vertical source array deployed in the far-field of a towed horizontal receiver array during the Gulf of Maine 2006 Experiment. The source transmitted intermittent broadband pulses in the 300-1200 Hz frequency range. All four methods are found to be comparable with averaged error of between 9% to 13% in estimating the mean source position in a wide variety of source-receiver geometries. In the case of a relatively stationary source, the synthetic aperture tracking outperforms the other three methods by a factor of two with only 4% error. For a moving source, the Kalman filter method yields the best performance with 8% error. The array invariant is the best approach for localizing sources within the endfire beam of the receiver array with less than 10% error.
Defusing the controversy of scattering versus diffusion coefficients
Authors: Gideonse Hendrik D.
Published: September 2012

Several methodologies for the testing and analysis of diffusers have been developed including the ISO Scattering Coefficient and the AES Diffusion Coefficient. These coefficients are the source of some controversy today and this paper makes the attempt to investigate the benefits and weaknesses of these tools by using them to research and test a new diffuser shape. Several issues are exposed in using the coefficients as both qualitative and quantitative metrics. The most important of these being problems with the validity of the comparison of the diffuser's behavior to that of a like-sized flat panel. Additionally there appears to be an intuitive disconnect between the perceived diffusive merits shown by polar plots and the numerical value of coefficients derived from the plots.
Architectural acoustics in educational facilities: An empirical study on university classrooms in Egypt
Authors: Awad Hadia S. , Farag Hania H. , Taha Dina S. , Hanafi Mohamed A.
Published: September 2012

There is a lack of acoustic performance standards for educational facilities in Egypt, especially university classrooms, resulting in a very bad acoustical quality. There is also a lack in the scientific research in that field of study in Egypt, although this type of research has been held in many other countries. This paper aims at studying the acoustic performance standards in different countries such as (ANSI-S12.60, 2010, BB93, 2003), and then using them to evaluate the acoustical quality in classrooms in Alexandria, including public and private, new and old constructions. The evaluation was done by three methods; B&K2250 SLM, Computer modeling using Odeon acoustic prediction software, and mathematical calculations using Sabine and Eyring formulae. The empirical study results were then compared to the recommended values by the previously mentioned standards. The study focused on sound levels, noise levels, reverberation time and Signal-to-Noise ratios. The results prove that there is a lack of acoustic design in Egypt. Most of the studied classrooms did not meet the recommended values. The main problem is using inappropriate finishing materials, which provide very high reverberation. Some simple and applicable proposals were suggested to improve the acoustical quality of classrooms. Predictions have shown a significant improvement.
Sound power measurements in non-ideal enclosures using acoustic energy density
Authors: Marquez Daniel R. , Sommerfeldt Scott D. , Leishman Timothy W. , Blotter Jonathan
Published: September 2012

Sound power measurements are generally made in reverberation or anechoic chambers using acoustic pressure measurements as outlined in specific ISO or other standards. Reverberation chambers are used to approximate diffuse acoustic fields wherein the sound power is directly proportional to the spatially averaged squared pressure. Anechoic chambers are utilized to create a direct field condition, wherein sound power can also be determined from sound pressure levels located on a measurement surface which envelopes the source. This paper will introduce a method that utilizes acoustic energy density to estimate the sound power produced in non-ideal enclosures when both direct and reverberant energies contribute significantly to the total acoustic field. Since the acoustic energy density in an enclosure is more spatially uniform than the acoustic pressure, this method can achieve the same accuracy in determining sound power with fewer measurement positions when spatially averaging. The results from numerical models of several rectangular rooms of varying acoustical properties will be presented and the accuracy of the method will be addressed
The effect of building reflections on the equivalent noise level (Leq) from traffic on Lake Shore Drive
Authors: McGowan Eric W.
Published: September 2012

The objective of this study was to determine if the Leq resulting from vehicular traffic on Lake Shore Drive in Chicago was affected by reflections from buildings being present in the background. Three test locations were chosen along the same stretch of road, two with buildings in the background, one without. A noise propagation model using CadnaA was constructed in order to compare results from the test locations. Traffic noise spectra were recorded at each location for one (1) hour and post-processed using SpectraPLUS; test data were also acquired using a Type II SPL meter with Leq capabilities. The location with the most building coverage in the background yielded the highest Leq, while the location with no buildings gave the lowest level. The computer model produced results that followed the trends of the test, but the predicted values were consistently higher than the measured levels. The study analyzed if the effect of parameters such as pavement and country-specific Standards could account for the differences between measured and modeled data.
Computer simulation of Benjamin Franklin's acoustic experiment on George Whitefield's oratory
Authors: Boren Braxton B. , Roginska Agnieszka
Published: September 2012

The Anglican preacher George Whitefield was renowned for his loud voice and the huge crowds he drew during the transatlantic revivals of the 18th century. Benjamin Franklin was skeptical of the accounts of crowds of 30,000 gathering in London, and when Whitefield came to Philadelphia in 1739, Franklin performed one of the earliest recorded ‘archeoacoustic’ experiments: walking backwards down Market Street, Franklin continued listening to Whitefield speak from the old courthouse until his sermon became unintelligible. Using this maximum intelligible distance, Franklin calculated that Whitefield probably could have been heard by more than 30,000 listeners. Using Franklin’s account and period maps and prints of the colonial city, we have built a virtual CAD model of Philadelphia as it would have existed during Whitefield’s visit. This paper discusses techniques employed using geometric acoustic simulation software to approximate the loudness of Whitefield’s voice based on the STI at Franklin’s position. To determine the STI, the background noise at Franklin’s position is simulated according to his account of a noise source on Front Street. Given a specific noise source and a minimum intelligible STI, this system yields a loudness value in dB-SPL for an acoustic source at Whitefield’s position.
Effects of short noise bursts on human performance and perception
Authors: Ainley Christopher , Wang Lily M.
Published: September 2012

The goal of this research project is to better quantify human reactions to short bursts of noise, to complement research at NASA Langley Research Center on evaluating human response inside buildings to low-level sonic booms. The project involved exposing participants over 30-minute sessions to 250 ms broadband noise bursts of certain levels, presented in a controlled yet randomized fashion throughout the session, and gathering responses on human perception and performance on an arithmetic task dealing with short-term memory. While previous research has demonstrated effects of noise bursts of varying amplitudes on other types of tasks that study cognitive processing including attention and at louder levels on this arithmetic task (i.e. 100 dB peak), more information is needed to indicate at what level and to what degree such noise bursts may impact human performance and perception. Twenty-seven test subjects were tested over multiple sessions, with four different levels of the noise bursts, ranging from 47 to 77 dBA. The gathered performance and perception results will be presented, with consideration of each subject’s self-reported noise sensitivity. The results will also be correlated to a number of noise metrics, including perceived level (PL) and L1-L99. [Work supported by a NASA Nebraska Space Grant.]
Whoosh noise measurements from an automotive centrifugal compressor
Authors: Dehner Rick , Figurella Neil , Fogarty Kevin , Selamet Ahmet
Published: September 2012

Broadband noise, accompanying the flow separation and turbulence, is studied for an automotive centrifugal compressor on a steady-flow turbocharger bench facility. When operated in the mid-flow region, the current compressor exhibited elevated broadband noise at high frequencies, which was evident both in the upstream compressor duct and external sound pressure level (SPL) measurement locations. Viewing SPL data as a function of the flow angle relative to the leading edge of the inducer blades (incidence angle) reveals a relationship which is nearly independent of the rotational speed. As the incidence angle is increased, broadband noise levels first go up gradually, nearly level off, and then decrease sharply at a critical incidence angle.
Progression of age-related voice instabilities in a non-pathological voice
Authors: Hunter Eric J. , Titze Ingo R.
Published: September 2012

Age-related structural and functional changes to the aerodigestive tract can affect breathing, swallowing, and voice. Not only can these changes can shape an individual’s quality of life, they can, ultimately, be life-threatening. Looking at the voice specifically, changes to the subglottal and supraglottal airways influence vocal fold vibration by producing major bifurcations in the vibration regime. These bifurcations are evidenced in the voice by aphonic segments, subharmonic or side-band frequencies, frequency jumps, and chaotic vibration. The increased occurrence of these bifurcations can, in turn, may indicate age related changes in the vocal folds. The current study examines age-related changes in voice production in two individuals spanning 48–98 y/o and 52–90 y/o. Previous studies revealed changes in breath rate and pitch begin between the ages of 68–74 y/o, indicating a fundamental change in the body’s maintenance of the speech mechanism. Voice breaks and bobbles are shown to correspond with this change. These voice breaks, along with the increased voice pitch previously reported, may indicate an interaction of the subglottal and supraglottal airway and an increased weakness of the vocal folds. Weakened vocal folds could also indicate compromised swallowing and airway protection mechanisms.
Variance in tongue motion patterns during the production of /s/
Authors: Ding Cindy , Woo Jonghye , Chen Hegang , Stone Maureen
Published: September 2012

One of the issues in speech motor control is the nature of variance that occurs in the production of a single speech task spoken by multiple speakers. Do they all essentially use the same gesture modified by fine tuning to adjust to their own anatomy, dialect, etc, or are there quite different, motor equivalent, ways to produce the same speech sound. Production of /s/ in American English is known to be produced using two methods: apical or laminal. Apical /s/ primarily elevates the tongue tip, while laminal /s/ utilizes the tip and blade. Both gestures are found frequently in normal speakers. The present study uses principal components analysis of midsagittal velocity fields to identify the patterns of variance in the internal tongue motion patterns of 10 normal speakers. Palate height will also be examined, as preliminary evidence points to low-palate speakers having a preference for apical /s/, while high-palate speakers use either. Tagged-MRI was used to record ‘a geese’, and the motion between /g/ and /s/ studied for amount and directions of variance. The goal is to identify stable features of the motion and the effects of /s/ type and palate height.
Single motor unit activity in the genioglossus muscle during vowel articulation
Authors: LaCross Amy , Takano Sayoko , Kidder Ian J. , Watson Peter J. , Bailey Elizabeth Fiona
Published: September 2012

There is an abundance of previous electromyographic (EMG) research conducted in the human tongue that has as its focus the extrinsic tongue muscle, genioglossus (GG) and its role in preserving upper airway patency for purposes of gas exchange. Comparatively few studies have documented GG EMG activities in the performance of volitional tasks such as speech production (Honda 1992; Honda et al. 1992; Kusakawa et al 1993). Here we report on de novo efforts to characterize GG single motor unit (SMU) activities in the context of volitional i.e., speech and automatic i.e., central pattern generator driven respiratory activities. Using tungsten microelectrodes inserted into the belly of the GG in 5 healthy adults, we recorded SMU activity during static articulation of the vowels [a] and [ae] in three conditions: phonated, whispered, and articulation using an electrolarynx. Preliminary findings provide evidence of recruitment and de-recruitment of GG motor units coincident with the onset and offset of vowel articulation in phonated and whispered, but not electrolarynx conditions. Furthermore, fluctuations in GG MU firing rates mirror intensity variations within phonated and whispered utterances. These findings provide much-needed new insights into the differential modulation of lingual motor unit activities for purposes of speech production versus breathing.
On parameterizing glottal area waveforms from high-speed images
Authors: Chen Gang , Kreiman Jody , Gerratt Bruce R. , Shue Yen-Liang , Alwan Abeer
Published: September 2012

Because voice signals result from vocal fold vibration, perceptually-meaningful vibratory measures should quantify those aspects of vibration that correspond to differences in voice quality. In this study, glottal area waveforms were calculated from high-speed images of the vocal folds. Principal component analysis was applied to these waveforms to investigate the factors that vary with voice quality. Results showed that the first two principal components were significantly (p < 0.01) associated with the open quotient and the ratio of alternating-current to direct-current components. However, these conventional source measures, which are based on glottal flow, do not fully characterize observed variations in glottal area pulse shape across different glottal configurations, especially with respect to patterns of glottal closure that may be perceptually important. A source measure, the Source Dynamic Index (SDI), is proposed to characterize glottal area waveform variation for both complete and incomplete glottal closures. Analyses of “glide” phonations in which quality varied continuously from breathy to pressed showed that the SDI is able to characterize the corresponding continuum of glottal area waveform variation, regardless of the presence or absence of glottal gaps. [Work supported in part by NSF and NIH.]
Perceptual evaluation of voicing source models
Authors: Kreiman Jody E. , Gerratt Bruce R. , Chen Gang , Garellek Mark , Alwan Abeer
Published: September 2012

Many models of the glottal source have been proposed, but none has been systematically validated perceptually, so that it is unclear whether deviations from perfect fit have perceptual importance. If model fit fails in ways that have no perceptual significance, such “errors” can be ignored, but poor fit with respect to perceptually-important features has both theoretical and practical importance. To address this issue, we fit 6 different source models to 40 natural voice sources, and then evaluated fit with respect to time-domain landmarks on the source waveforms and details of the harmonic voice source spectrum. We also generated synthetic copies of the voices using each modeled source pulse, with all other synthesizer parameters held constant, and then conducted a visual sort-and-rate task in which listeners assessed the extent of perceived match between the original natural voice samples and each copy. Discussion will focus on the specific strengths and weaknesses of each modeling approach for characterizing differences in vocal quality. [Work supported by NIH/NIDCD grant DC01797 and NSF grant IIS-1018863.]
Intraglottal velocity and displacement measurements in an excised larynx
Authors: Oren Liran , Gutmark Ephraim , Khosla Sid
Published: September 2012

A major assumption of previously published PIV measurements in excised larynges, is that the vortices seen directly above the glottal exit during closing are due to the flow separation vortices (FSV) that formed in the glottis. This assumption needed experimental validation. In addition, the pressures associated with the vortices above the vocal folds may be different than the pressures generated by the intraglottal FSV. Previous studies also relied on images of the glottal opening taken from above of the folds to evaluate the displacement of the folds. This method cannot separate the dynamic of the folds that occurs along their superior and inferior edges. The current study uses a major modification of the PIV techniques previously described to simultaneously measure intraglottal velocity fields and intraglottal geometry. The intraglottal pressure distribution is computed from the velocity measurements by solving the pressure Poisson equation. The results show that strong negative pressure is formed towards the superior edge of the folds. This negative pressure can produce additional force during closing. The displacement of the folds, during closing, is also extracted from the PIV images and it shows that the acceleration of the superior edge is consistently higher than the inferior edge.
Intraglottal pressures related to glottal and laryngeal asymmetries
Authors: Scherer Ronald
Published: September 2012

The air pressures within the glottis during phonation may be highly dependent upon the symmetry of the glottis and other laryngeal structures. Physical laryngeal models M5 and M6 have been used to explore the slanted “oblique” glottis with different included and oblique angles, the change to three dimensions, and the presence of the arytenoid cartilages. Results suggest that (1) the greater the oblique angle, the higher the glottal entrance pressures compared to the symmetric glottis, (2) the larger the intraglottal angle, the more different the pressures are on the two glottal sides, (3) pressures are higher on the convergent side than on the divergent side of the glottis, (4) intraglottal pressure depends on which side of the glottis the flow exits, with the Flow Wall side having lower pressures, (5) for large glottal diameters, all intraglottal pressures may be lower on the Flow Wall side, (6) for special cases of high transglottal pressure, the pressure near glottal exit may be lower than at entrance (divergent glottis), (7) a symmetric but 3-dimensional glottis has consistent but minor pressure changes in the anterior-posterior direction, and (8) the presence of the arytenoid cartilages has minor effects on the intraglottal pressures. [Support from NIH.]
An articulatory and acoustic study of fricative consonants /s/ and /sh/ in normal and post-glossectomy speakers
Authors: Zhou Xinhui , Jonghye Woo , Stone Maureen , Espy-Wilson Carol Y.
Published: September 2012

Glossectomy is a surgical procedure to remove the cancerous tumor of the tongue. After the glossectomy, the tongue is sutured closed or a flap is inserted to reconstruct the tongue volume. As a result, the properties of the tongue are more or less affected by the surgery. The changes in the tongue properties may also affect the speech production abilities of the post-glossectomy speaker. This study examined the production of the fricative consonants /s/ and /sh/ in normal and post-glossectomy speakers. The data analyzed consisted of audio and magnetic resonance images from dozens of normal and glossectomy speakers. An acoustic analysis showed that the average centers of gravity of /s/ and /sh/ in glossectomy speakers are significantly lower than in normals. This difference may be explained by a more posterior constriction in glossectomees due to the surgery. Examination of the tongue shapes in midsagittal MR images showed that they tend to have more laminal /s/ than apical /s/. 3-D vocal tracts of /s/ and /sh/ were reconstructed for three glossectomy speakers whose /s/ and /sh/ cannot easily be discriminated in listening tests. Details of the 3-D vocal tract shapes, along with their acoustic implications, will be discussed for the glossectomy and normal speakers. [This research was supported by NIH R01 CA133015.]
Acoustic discrimination of Parkinsonian speech using cepstral measures of articulation
Authors: Skowronski Mark D. , Shrivastav Rahul , Harnsberger James , Anand Supraja , Rosenbek Jay
Published: September 2012

The effects of idiopathic Parkinson’s disease (IPD) on speech include articulatory imprecision. We quantified articulation rate and range acoustically using cepstral coefficients to represent vocal tract settings. Cepstral coefficients were extracted from 10 sentences spoken by 76 talkers, half of which were diagnosed with IPD. Articulation range was estimated for each sentence as the sum across cepstral coefficients of the standard deviation of each coefficient, and articulation rate was estimated using the same procedure, replacing cepstral coefficients with delta coefficients. The mean ± standard deviation (N = 380 sentences) for the articulation measures of range (7.95 ± 0.50 vs. 6.66 ± 0.53) and rate (5.64 ± 0.56 vs. 4.40 ± 0.46) were significantly different (t-test, p < 0.001) for normal vs. IPD speech, respectively. In a leave-one-talker-out classification experiment, range accuracy was 90.1%, rate accuracy was 88.8%, and accuracy was 92.9% using a combined model of articulation range and rate. The strengths of the articulation measures include 1) sensitivity to IPD speech, 2) reliance on cepstral coefficients which have been used for over 30 years to represent speech, 3) no segmentation requirements, 4) low sensitivity to speech material, and 5) effective with only 2 seconds of speech.
Second-formant locus patterns in dysarthric speech
Authors: Kim Heejin , Hasegawa-Johnson Mark
Published: September 2012

Second-formant (F2) locus equations represent a linear relationship between F2 measured at the vowel onset following stop release and F2 measured at the vowel midpoint in a consonant-vowel (CV) sequence. Prior research has used the slope and intercept of locus equations as indices to coarticulation degree and the consonant’s place of articulation. This presentation addresses coarticulation degree and place of articulation contrasts in dysarthric speech, by comparing locus equation measures for speakers with cerebral palsy and control speakers. Locus equation data are extracted from the Universal Access Speech (Kim et al. 2008). The data consist of CV sequences with labial, alveolar, velar stops produced in the context of various vowels that differ in backness and thus in F2. Results show that for alveolars and labials, slopes are less steep and intercepts are higher in dysarthric speech compared to normal speech, indicating a reduced degree of coarticulation in CV transitions, while for front and back velars, the opposite pattern is observed. In addition, a second-order locus equation analysis shows a reduced separation especially between alveolars and front velars in dysarthric speech. Results will be discussed in relation to the horizontal tongue body positions in CV transitions in dysarthric speech.