bluuebunny/crossref_metadata_2025 huggingface dataset:
On learning to recognize spectrally reduced speech: I. The effects of multiple repetitions
Authors: Eng Michelle , Chiu Peter
Published: October 2013

In a previous study [Chiu et al., J. Acoust. Soc. Am. 109, 2501 (2001)], normal listerers learned open-set recognition of words in Harvard IEEE sentences that were spectrally reduced to 4 channels using the CIS processing strategy [Shannon et al., J. Acoust. Soc. Am. 104, 2467–2476 (1998)]. Half of the listeners listened to a male talker during training, the other half a female talker. At different points in the training listeners were tested with blocks of IEEE sentences spoken by the familiar talker, blocks of IEEE sentences by the novel talker, blocks of IEEE sentences of the 2 talkers randomly intermixed, and blocks of HINT sentences. In that study, listeners could listen to each sentence during early training for an unrestricted number of times before they were given feedback. In the current study, listeners were allowed to listen to each sentence only once during training before feedback was given. Preliminary data obtained from the current study were largely comparable to those obtained in the previous study. The results suggest that massed repetitions and drills, at least in the context of our experimental conditions, wa have minimal effect on learning to recognize spectrally reduced speech. [Work supported by URC, Univ. of Cincinnati.]
The effects of room acoustics and multiple talkers on speech intelligibility with normal hearing and hearing impaired individuals
Authors: Mishler Pamela , Ericson Mark , Cowell Shawn
Published: October 2013

Many studies have shown the effect of hearing loss and spatial separation on multi-talker word identification ability in free-field listening conditions. However, little is known about the combination of room reflections and hearing impairment on understanding speech. The purpose of this study was to determine if the effect of room reflections on speech intelligibility was equal among normal hearing and hearing impaired listeners. Two acoustic generation techniques were employed to create the free-field and semi-reverberant listening conditions. Free-field listening conditions were created using virtual acoustic technology. Room reflections were added to the speech samples using dummy-head recordings in a semi-reverberant room. The word identification test used in the experiments was the Modified Rhyme Test (MRT), which afforded a multitude of phonemic pairs when phrases were played simultaneously. Thirty-three listeners in each group participated in each test condition. The hearing impaired listeners were below 60 years of age and exhibited a high frequency, cochlear hearing loss. The normal hearing listeners were age matched with the hearing impaired group. A difference in the type of phonemic identifications between the two techniques and the two groups were found. These phonemic contrasts between will be discussed individually.
Relationships among vocabulary size, nonverbal cognition, and spoken word recognition in adults with cochlear implants
Authors: Collison Elizabeth A. , Munson Benjamin , Carney Arlene E.
Published: October 2013

Recent research has attempted to identify the factors that predict speech perception performance among users of cochlear implants (CIs). Studies have found that approximately 20%–60% of the variance in speech perception scores can be accounted for by factors including duration of deafness, etiology, type of device, and length of implant use, leaving approximately 50% of the variance unaccounted for. The current study examines the extent to which vocabulary size and nonverbal cognitive ability predict CI listeners’ spoken word recognition. Fifteen postlingually deafened adults with nucleus or clarion CIs were given standardized assessments of nonverbal cognitive ability and expressive vocabulary size: the Expressive Vocabulary Test, the Test of Nonverbal Intelligence-III, and the Woodcock-Johnson-III Test of Cognitive Ability, Verbal Comprehension subtest. Two spoken word recognition tasks were administered. In the first, listeners identified isophonemic CVC words. In the second, listeners identified gated words varying in lexical frequency and neighborhood density. Analyses will examine the influence of lexical frequency and neighborhood density on the uniqueness point in the gating task, as well as relationships among nonverbal cognitive ability, vocabulary size, and the two spoken word recognition measures. [Work supported by NIH Grant P01 DC00110 and by the Lions 3M Hearing Foundation.]
The contribution of auditory temporal processing to the separation of competing speech signals in listeners with normal hearing
Authors: Adam Trudy J. , Pichora-Fuller Kathy
Published: October 2013

The hallmark of auditory function in aging adults is difficulty listening in a background of competing talkers, even when hearing sensitivity in quiet is good. Age-related physiological changes may contribute by introducing small timing errors (jitter) to the neural representation of sound, compromising the fidelity of the signal’s fine temporal structure. This may preclude the association of spectral features to form an accurate percept of one complex stimulus, distinct from competing sounds. For simple voiced speech (vowels), the separation of two competing stimuli can be achieved on the basis of their respective harmonic (temporal) structures. Fundamental frequency (F0) differences in competing stimuli facilitate their segregation. This benefit was hypothesized to rely on the adequate temporal representation of the speech signal(s). Auditory aging was simulated via the desynchronization (∼0.25-ms jitter) of the spectral bands of synthesized vowels. The perceptual benefit of F0 difference for the identification of concurrent vowel pairs was examined for intact and jittered vowels in young adults with normal hearing thresholds. Results suggest a role for reduced signal fidelity in the perceptual difficulties encountered in noisy everyday environments by aging listeners. [Work generously supported by the Michael Smith Foundation for Health Research.]
Reverberation reduction by sub-band unmasking
Authors: Libbey Brad , Rogers Peter
Published: October 2013

Reverberation reduces the intelligibility of speech recorded in rooms. Signal processing algorithms attempt to reduce this reverberation such that the speech not only sounds less reverberant, but is more intelligible. Historically, these techniques are based on calculations of the room impulse response, on arrays of two or more microphones, noise cancellation, or cepstral techniques. The new technique assumes that the intelligibility loss is due to overlap masking of reverberant tails of speech. The algorithm looks at sub-bands and identifies those that mask adjacent bands. When this occurs in conjunction with a decaying envelope the decay rate is increased. Thus, the masked band has a greater chance of being heard. Intelligibility results for the processed speech will be presented.
Recognition masking for transitions in initial and final position
Authors: Gaston Jeremy R. , Pastore Richard E.
Published: October 2013

There is a common implicit assumption that the input to higher or specialized processes is accurately represented by the physical properties of the stimulus, with, at most, only detection masking and some basic initial filtering. Since playing an auditory stimulus backwards does not alter the spectral composition of the stimuli, this initial assumption has led to the conclusion that any differences in perception must be attributed to higher level or specialized processes. Thus, for example, significant differences in the perception of temporally reversed CV syllables are assumed to reflect only specialized processes (i.e., speech module). The current study challenges this basic implicit assumption. Backward ‘‘recognition’’ masking is a basic, but poorly understood perceptual phenomenon in which an added following stimulus interferes with discrimination of simple preceding tones. Recent work from our laboratory indicates similar interference in the recognition of easily discriminated chords. The current study investigates the role of backward recognition masking in determining ability to recognize the direction of frequency glides similar to formant transitions of stop consonants.
Acoustical considerations for an automotive hands-free mobile terminal
Authors: Ruhala Richard J. , Iyengar Vasu
Published: October 2013

The harsh acoustical environment of an automotive interior yields significant difficulty in controlling acoustic echo and noise during ‘‘hands-free’’ wireless phone calls. In addition to having multiple closely coupled reflection paths, the user (talker) is often positioned between the loudspeaker and microphone path. This can cause frequent changes in the acoustic path which necessitates fast and accurate adaptation of the acoustic echo canceller and/or non-linear controller. Further, the frequency response between the talker and the microphone is altered due to cabin resonances and modes; it is also strongly affected by their positions. Additional acoustical considerations for automotive hands-free implementations are the high levels and non-stationary statistics of background noise, high amplification, and possible resultant distortion of the received audio signal, and non-linear network echo. These impairments yield significant challenges in controlling the acoustic echo and noise in full-duplex, partial-duplex, and even half-duplex systems. In this work, impulse response, frequency response, and SNR measurements are presented and analyzed to demonstrate the critical differences between a traditional desktop speakerphone and an automotive hands-free mobile terminal.
Acoustic and perceptual effects of overall F0 range in a lexical pitch accent distinction
Authors: Wade Travis
Published: October 2013

A speaker’s overall fundamental frequency range is generally considered a variable, nonlinguistic element of intonation. This study examined the precision with which overall F0 is predictable based on previous intonational context and the extent to which it may be perceptually significant. Speakers of Tokyo Japanese produced pairs of sentences differing lexically only in the presence or absence of a single pitch accent as responses to visual and prerecorded speech cues presented in an interactive manner. F0 placement of high tones (previously observed to be relatively variable in pitch contours) was found to be consistent across speakers and uniformly dependent on the intonation of the different sentences used as cues. In a subsequent perception experiment, continuous manipulation of these same sentences between typical accented and typical non-accent-containing versions were presented to Japanese listeners for lexical identification. Results showed that listeners’ perception was not significantly altered in compensation for artificial manipulation of preceding intonation. Implications are discussed within an autosegmental analysis of tone. The current results are consistent with the notion that pitch range (i.e., specific vertical locations of tonal peaks) does not simply vary gradiently across speakers and situations but constitutes a predictable part of the phonetic specification of tones.
The effect of variation in naturalness on phonetic perceptual identification
Authors: Remez Robert E. , Yang Cynthia Y. , Piorkowski Rebecca L. , Wissig Stephanie , Batchelder Abigail , Nam Heddy
Published: October 2013

The relation between apparent naturalness and phonetic identification was assessed in six perceptual tests. A seven-step place-of-articulation series spanning [da] to [ga] was created with speech synthesis approximating the spectra of natural samples. The sensitivity of perceivers to this realization of a place contrast was assessed by estimating the cumulative d′ across the series in identification tests. Four variants of this series differing in apparent naturalness were produced by altering the synthesis source function while preserving the center frequency and bandwidth of the formants, and by replicating the gross spectrotemporal patterns with time-varying sinusoids. In addition to calibrating perceivers’ sensitivity to the place contrast over variation in naturalness, we conducted a naturalness tournament composed of items drawn from the five test series. A correlation of the findings of the naturalness tournament with the measures of phonetic sensitivity offers an index of the effect of variation in naturalness on phonetic perception. This study can resolve the dispute between the classic premise that intelligibility and naturalness are orthogonal attributes of speech perception, and the more recent premise entailed by episodically based accounts of perceptual categorization, that novel instances are identified by virtue of auditory similarity to prior exemplars. [Research supported by NIDCD.]
Signal detection analysis of response distributions for intensity and speech judgments
Authors: Pastore Richard E. , Macmillan Neil A.
Published: October 2013

Schouten and van Hessen [J. Acoust. Soc. Am. 104, 2980–2990 (1998)] reported rating response distributions for intensity and speech continua. These empirical distributions were claimed to reflect each subject’s internal decision axes, allowing direct evaluation of the distribution assumptions underlying the common Gaussian equal-variance model signal detection theory (SDT). The observed distributions were deemed roughly Gaussian. However, the original analyses implicitly assume that the rating scale is a direct, interval reflection of internal likelihood values. We used the data instead to construct ROC curves, an analysis that assumes an ordinal relationship between the rating responses and the decision axes. Our ROC analyses show that the representations of the intensity stimuli reasonably approximate a Gaussian model. In contrast, the distributions underlying the speech stimuli are clearly not Gaussian, but imply discrete representations. We discuss appropriate decision models for judgments of intensity and speech continua.
Elderly perception of speech from a computer
Authors: Black Alan , Eskenazi Maxine , Simmons Reid
Published: October 2013

An aging population still needs to access information, such as bus schedules. It is evident that they will be doing so using computers and especially interfaces using speech input and output. This is a preliminary study to the use of synthetic speech for the elderly. In it twenty persons between the ages of 60 and 80 were asked to listen to speech emitted by a robot (CMU’s VIKIA) and to write down what they heard. All of the speech was natural prerecorded speech (not synthetic) read by one female speaker. There were four listening conditions: (a) only speech emitted, (b) robot moves before emitting speech, (c) face has lip movement during speech, (d) both (b) and (c). There were very few errors for conditions (b), (c), and (d), but errors existed for condition (a). The presentation will discuss experimental conditions, show actual figures and try to draw conclusions for speech communication between computers and the elderly.
Patterns in the perception of VC(C)V strings
Authors: Nearey Terrance M. , Smits Roel
Published: October 2013

A set of 144 /aC(C)a/ stimuli were constructed (inspired by, e.g., B. Repp, Percept. Psychophys. 24, 471–485 (1978)]). Each consisted of: a vocoid (135 ms steady state followed by 50 ms VC closing transitions); a silent portion; and another vocoid (50 ms CV opening transition plus 250 ms steady state). Four different silent periods [80, 120, 190, and 300 ms] were crossed with six closing transition patterns (ranging from b-like to d-like closures) and with six analogous opening transitions. Thirteen listeners classified each stimulus ten times. Polytomous logistic regression showed that singletons, heterorganic clusters, and geminates all have distinct duration weights. Contrary to expectations, results for place were remarkably simple. Weights for closing transitions show labial closures (/b, bd, bb/) to be widely separated from the coronal closures (/d, db, dd/), but are otherwise homogeneous. Weights for opening cues show a similar dichotomy of labial releases /b, db, bb/ versus coronal releases (/d, bd, dd/); here, however, the singleton /b/ differs slightly from the other labials. A related experiment in Dutch will be discussed and results from both experiments will be compared against proposals from the literature and our own models. [Work supported by SSHRC.].
The effect of monauralized binaural displays on speech intelligibility
Authors: Bolia Robert S.
Published: October 2013

It has often been demonstrated in laboratory settings that the spatial separation of multiple simultaneous talkers leads to enhanced intelligibility of a target message as well as an increase in the ease with which nontarget messages can be successfully ignored. This finding can be exploited in binaural intercom systems to improve communications’ effectiveness. In some situations, operators use monaural displays at least part of the time, for example when operators in command and control nodes remove one earcup to talk to other nearby operators, in an effort to avoid squandering channels. The present investigation was designed to determine whether such a temporary monauralization of a binaural intercom would result in reduced speech intelligibility relative to a diotic control condition. Participants listened to 2 or 3 simultaneous presentations of phrases from the coordinate response measure (Bolia, Nelson, Ericson, and Simpson, 2000) and responded according to the instructions spoken by the target talker. The dependent measure was the percentage of correct responses. Independent variables included listening condition (monaural left, monaural right, binaural) and the locations of the target and distractor talkers. Results indicate that the performance in the monaural conditions were worse than in the binaural conditions but better than the diotic control.
Importance of onset properties on spectral contrast for speech and other complex spectra
Authors: Coady Jeffry A. , Kluender Keith R.
Published: October 2013

Previous studies using speech and nonspeech analogs have shown that auditory mechanisms that enhance spectral contrast serve to compensate for assimilation of spectral composition caused by coarticulated production. In the present studies, series of CV syllables varying acoustically in F2-onset frequency and perceptually from /ba/ to /da/ were identified either following [e] (higher F2) and [o] (lower F2) or following spectral complements of these vowels. Spectral complements had troughs replacing peaks corresponding to formants in the vowels. Both preceding vowels and spectral complements altered perception of the following stop, with complementary spectra providing results complementary to those for vowel spectra. However, effects for preceding spectral complements differed from those for preceding vowels by being dependent upon the precursor duration or inter-stimulus interval (ISI). Additional experiments using short spectral samples revealed that the single most important predictor of effects of vowel complements was the interval between the onset of the preceding sound and the onset of the following CV. This finding appears to implicate spectral characteristics of onsets (in addition to or instead of simple adaptation or masking) as being important for spectral contrast. The significance of these findings for the perception of connected speech will be described. [Work supported by NIDCD.]
Speaking rate adjustment across changes in talker
Authors: Newman Rochelle S. , Sawusch James R.
Published: October 2013

Individuals vary their speaking rate, and listeners use the speaking rate of precursor sentences to adjust for these changes [Kidd (1989)]. Recent work has suggested that speaking rate adjustment may not always be limited to speech from a single talker (Sawusch and Newman, 2000). When a talker change occurs in the midst of a vowel, the durations of both portions of the vowel influence perception of the rate-based contrast. In the present experiments, we examined the effect of talker change on rate normalization for precursor phrases. A male talker produced the sentence, I heard him say the word—at one of three different speaking rates. A female talker then produced a nonword containing a duration-based contrast. We examined whether the male talker’s speaking rate would influence perception of the female talker’s speech. The results were somewhat surprising. The speaking rate of the first talker did influence perception of the contrast in the second talker. However, the effect was a U-shaped function of speaking rate, rather than the linear function typically demonstrated in the single-voice condition. Several follow-up studies replicated this basic pattern. Implications of this finding for studies of rate normalization will be discussed.
The effect of familiarity with a voice on continuous shadowing: Effects of explicit knowledge, but not of implicit familiarity
Authors: Newman Rochelle S.
Published: October 2013

Listeners were asked to shadow a target voice while a second voice spoke fluently in the background. Listeners differed in their familiarity with the target voice: members of the first group of listeners were familiar with the voice and were told explicitly whose voice they would be hearing; members of a second group were familiar with the target voice but were not warned whose voice it was, and members of a third group were entirely unfamiliar with the target voice. Familiarity was from real-world experience: the voice was that of one of the participant’s professors. Explicit knowledge of talker identity appeared to have the larger effect on listener performance: Participants in the first group made significantly fewer shadowing errors than those in the second group. In contrast, simple familiarity with a voice had no such influence. Familiarity did influence the types of errors listeners made, however: those participants who were familiar with the target voice prior to the test session made fewer incorrect responses than did those who had not heard the talker previously, although their total number of errors (including misses, as well as incorrect responses) did not differ.
Talker identification from analysis of raw complex waveforms
Authors: Stokes Michael A.
Published: October 2013

Stokes (1996) demonstrated that visual inspection of raw complex waveforms can be used to identify a vowel produced by a talker. This research resulted in the MAS Model of Vowel Perception and Production (Stokes, 1998; http://home.indy.net/∼masmodel/). More recently, another experiment extended this work to female talkers as well as male talkers (Stokes, 2001). Together, this research represents the only ongoing comprehensive research involving visual inspection of raw complex waveforms for identifying vowels produced by any talker. As an extension of the work, the present study involves identifying a talker from a waveform display. Unique voice signatures identified from waveform displays are used to identify a talker from a set of 10 talkers in the same way as one would identify a person from fingerprints. In two trials (the word who’d in trial 1 and heed in trial 2), a talker was correctly identified from a set of 10 unique talkers per trial using small visual samples of waveforms and matching it to a waveform sample of the talkers to be identified.
Mapping the perceptual space defined by transition onset frequencies of synthetic CV stimuli
Authors: Hawks John W. , Wead Emily , Fourakis Marios
Published: October 2013

The perception of formant transition cues to place of articulation was investigated through the use of synthetic stop-vowel stimuli. Three sets of stop-vowel stimuli were synthesized. In each set, the F2 and F3 onset frequencies were varied in an orthogonal design resulting in 110 combinations. The sets differed in the F2 and F3 transition offset frequencies which were set to be appropriate for the vowels [i, u, a]. Twenty normal-hearing listeners identified each stimulus as beginning with [b], [d] or [g]. Listeners were also asked to rate the certainty of their responses on a scale of 1 (least certain) to 5 (most certain). While resembling stop-vowel sequences, no burst or aspiration parameters were included, thus forcing listeners to use only transition information as the primary cue to place of articulation. Perceptual spaces were constructed as defined by the F2 and F3 onset frequencies. In each perceptual space each F2/F3 onset combination was assigned the phonetic code [b], [d], or [g], reflecting the plurality of subjects’ responses with certainty ratings higher than 3. The resulting spaces will be discussed in terms of the effect of the target vowel on the perception of F2 and F3 transitions.
Broadband ocean reverberation coherence
Authors: Reeves Jon , Morella Gary
Published: October 2013

Utilizing a wide-band transmitting/receiving system, at-sea experimental results for broadband reverberation have been acquired for both a harsh as well as a moderate ocean acoustic environment. Both stationary and moving sonar systems were employed in the study. Replica correlation (so-called match filter) processing of linear FM (LFM) pulse codes is employed to study the degree of reverberation coherence achievable over 4 octaves of bandwidth. The LFM signals are systematically processed over a bandwidth interval from 0.5 kHz to over 12 kHz.The corresponding reverberation coherence is displayed in terms of the variance of the cross correlation coefficient as a function of time-bandwidth product. Comparison with the theoretically expected variance for band-limited white Gaussian noise is also included. [Work sponsored by L. Jacobi, ONR 333.]
Localization when the source function is known
Authors: Fialkowski Laurie T. , Lingevitch Joseph F. , Orris Gregory J. , Nicholas Michael , Perkins John S.
Published: October 2013

Frazer and Pecholcs [J. Acoust. Soc. Am. 88, 995–1002 (1990)] have investigated single-hydrophone time-domain localization. They introduced several localizers, similar to an earlier method introduced by Clay [J. Acoust. Soc. Am. 81, 660–664 (1987)], that differ in the amount of a priori information required about the source function. In this paper we revisit these localizers to look at the special case where the source function is known and explore the benefits of multiple hydrophones. This case is of interest due to its similarity to active matched field processing and because these localizers can be used as cost functions in environmental inverse problems when the source position is known precisely. We apply the localizers to data that includes several different types of source waveforms, and was collected in shallow water southwest of Key West, Florida. [Work supported by ONR.]
Challenges and solutions for realistic room simulation
Authors: Begault Durand R.
Published: October 2013

Virtual room acoustic simulation (auralization) techniques have traditionally focused on answering questions related to speech intelligibility or musical quality, typically in large volumetric spaces. More recently, auralization techniques have been found to be important for the externalization of headphone-reproduced virtual acoustic images. Although externalization can be accomplished using a minimal simulation, data indicate that realistic auralizations need to be responsive to head motion cues for accurate localization. Computational demands increase when providing for the simulation of coupled spaces, small rooms lacking meaningful reverberant decays, or reflective surfaces in outdoor environments. Auditory threshold data for both early reflections and late reverberant energy levels indicate that much of the information captured in acoustical measurements is inaudible, minimizing the intensive computational requirements of real-time auralization systems. Results are presented for early reflection thresholds as a function of azimuth angle, arrival time, and sound-source type, and reverberation thresholds as a function of reverberation time and level within 250-Hz–2-kHz octave bands. Good agreement is found between data obtained in virtual room simulations and those obtained in real rooms, allowing a strategy for minimizing computational requirements of real-time auralization systems.
Comparison of two speech privacy measurements, articulation index (AI) and speech privacy noise isolation class (NIC′), in open workplaces
Authors: Yoon Heakyung C. , Loftness Vivian
Published: October 2013

Lack of speech privacy has been reported to be the main dissatisfaction among occupants in open workplaces, according to workplace surveys. Two speech privacy measurements, Articulation Index (AI), standardized by the American National Standards Institute in 1969, and Speech Privacy Noise Isolation Class (NIC′, Noise Isolation Class Prime), adapted from Noise Isolation Class (NIC) by U. S. General Services Administration (GSA) in 1979, have been claimed as objective tools to measure speech privacy in open offices. To evaluate which of them, normal privacy for AI or satisfied privacy for NIC′, is a better tool in terms of speech privacy in a dynamic open office environment, measurements were taken in the field. AIs and NIC’s in the different partition heights and workplace configurations have been measured following ASTM E1130 (Standard Test Method for Objective Measurement of Speech Privacy in Open Offices Using Articulation Index) and GSA test PBS-C.1 (Method for the Direct Measurement of Speech-Privacy Potential (SPP) Based on Subjective Judgments) and PBS-C.2 (Public Building Service Standard Method of Test Method for the Sufficient Verification of Speech-Privacy Potential (SPP) Based on Objective Measurements Including Methods for the Rating of Functional Interzone Attenuation and NC-Background), respectively.
Analysis of the uncertainty budget for the reflection and absorption coefficient given by the two-microphone method
Authors: Esquivel D. Adolfo , Silva P. Guillermo
Published: October 2013

The method for measurement of the reflection and absorption coefficient at normal incidence consists of a sound source at one end of an impedance tube, where the plane waves propagated are reflected from a sample that is at the other end of the tube. Those plane waves are generated by a random signal, and the decomposition of the standing wave pattern is carried out measuring the acoustical pressure at two-fixed microphone positions, from which it is possible to determine a transfer function to obtain the reflection and absorption coefficient. The systematic errors associated with the two-microphone technique using an impedance tube are analyzed, i.e., microphone-sample distance, microphone spacer, atmospheric measurement conditions, correction factors to the microphone sensitivity, and phase and amplitude mismatch in the microphones. These parameters constitute some of the sources to the measurement uncertainty of the measurand-reflection and absorption coefficient. The estimation of the measurement uncertainty for the reflection and absorption coefficient by the two-microphone method; likewise the analysis of the uncertainties sources and its sensitivity coefficients are discussed using a linear approach model. Furthermore, the influence of the correction factor due to the frequency response mismatch in the microphones in the calculation of the reflection and absorption coefficient is experimentally evaluated.
Modeling strategies of ultrasound backscattering by blood
Authors: Guy Cloutier , David Savery , Isabelle Fontaine , Beng Ghee Teh
Published: October 2013

Tissue characterization using ultrasound (US) scattering can allow the identification of relevant cellular biophysical information noninvasively. The characterization of the level of red blood cell (RBC) aggregation is one of the proposed applications. Different modeling strategies have been investigated by our group to better understand the mechanisms of US backscattering by blood, and to propose relevant measurable indices of aggregation. It could be hypothesized from these studies that the microstructure formed by RBC clusters is a main determinant of US backscattered power. The structure factor, which is related to the Fourier transform of the microscopic density function of RBCs, is described and used to explain the scattering behavior for different spatial arrangements of nonaggregated and aggregated RBCs. The microscopic density function was described by the Percus–Yevick approximation (nonaggregated RBCs), and for aggregated RBCs, by the Poisson distribution, the Neyman–Scott point process, and very recently by a flow-dependent rheological model. These statistical and microrheological models allowed the study of US backscattered power as a function of the hematocrit, scatterers’ size, insonification frequency, and level of RBC aggregation. Experimental results available from the literature were used to validate the different approaches. [Work supported by Canadian Institutes of Health Research (MOP-36467), HSFQ, FCAR, and FRSQ.]
Coupled vibration analysis of large arterial vessels
Authors: Zhang Xiaoming , Fatemi Mostafa , Greenleaf James F.
Published: October 2013

A coupled vibration model, which takes into account the elastic arterial wall, the interior blood, and exterior tissue, is presented. The arterial wall is considered as a cylindrical elastic shell and is modeled with the thick shell theory due to the large thickness-to-radius ratios. Both the blood and the tissue are considered as ideal fluids and are modeled with acoustic wave theory. A newly developed method, the wave propagation approach, is extended for the coupled vibration of arterial walls. The theoretic model is evaluated against those available in the literature. The experiments were carried out on a silicone rubber tube in a water tank with an ultrasound stimulated system. This system uses the radiation force of ultrasound to vibrate the tube at low frequency and records the resulting response by a laser vibrometer. Both the excitation and measurement are remote and noncontact. The measured fundamental frequency is in good agreement with the theoretical prediction.
Measurement of stress/charge coefficients of a thin polyvinylidene fluoride film
Authors: Zahui Marcellin , Bibel George
Published: October 2013

A new experimental/analytical approach is used to measure a piezoelectric coefficient of polyvinylidene fluoride (PVDF). This method is based on the volume displacement of a vibrating beam. The obtained value of the stress/charge coefficient is checked using modal coordinate measurement theory and finite element analysis. A PVDF film is cut into two sensors shape. These sensors measure, respectively, the volume displacement and modal coordinate of a simply supported beam vibrating at its first mode. An accelerometer and a force gauge are used to measure the volume displacement and the modal coordinate of the beam. Then, PVDF volume displacement sensor equation developed by Guigou et al. is used with the experimental data to find the stress/charge coefficient e31. This value of e31 is first checked using the PVDF modal sensor equation from Lee and Moon. Then, the anys finite element code is used to validate the method. Experimental value of e31 obtained at room temperature is presented and compared with existing data. Excellent agreement is found. The extension of the method to 2D structure to determine the value of e31 and e32 simultaneously is introduced.
Background noise model of the piezoresistive microphone
Authors: Zuckerwar Allan J. , Kuhn Theodore R. , Serbyn M. Roman
Published: October 2013

A model has been developed to describe the sources of background noise in piezoresistive microphones. Three noise sources have been identified: mechanical Johnson noise due to the Brownian motion of air molecules impinging on the diaphragm, electrical Johnson noise in the bridge resistors, and generation–recombination noise in the bridge resistors. The model includes a set of lumped elements to represent the acoustic impedance of the diaphragm, another set of lumped elements to account for the stress across the bridge resistors, and a controlled source to yield an output voltage proportional to the stress due to the piezoresistive effect. Predictions based on the manufacturer’s data agree with experimental measurements. Representing the piezoresistive microphone as a nonreciprocal device, the model can be used to determine the microphone sensitivity and frequency response, as well as the background noise.
PC-based real-time acoustic source locator and sound capture system for teleconferencing
Authors: Morde Ashutosh , Grove Deborah , Utama Robert
Published: October 2013

A PC-based real time acoustic source locator and sound capture system has been developed. The system is implemented using Frontier Design A/D converters and the Intel Signal Processing Library directly on a 1 GHz Pentium III machine, without a DSP board. The source locator uses the cross-power spectral phase to locate a moving talker. The algorithm also uses an energy detector that minimizes incorrect location estimates by neglecting frames with high background noise. The source locator provides 8 location estimates per second. A 16-element 0.90 m linear delay-sum beamformer has also been implemented in the system as a method for selective sound capture. The ability of the source locator to detect talkers in a typical office environment is evaluated. In addition, the array response is measured. [Work supported by Intel.]
Independent component analysis for onset detection in piano trills
Authors: Brown Judith C. , Todd Jeremy G. , Smaragdis Paris
Published: October 2013

The detection of onsets in piano music is difficult due to the presence of many notes simultaneously and their long decay times from pedaling. This is even more difficult for trills where the rapid note changes make it difficult to observe a decrease in amplitude for individual notes in either the temporal wave form or the time dependent Fourier components. Occasionally one note of the trill has a much lower amplitude than the other making an unambiguous determination of its onset virtually impossible. We have analyzed a number of trills from CD’s of performances by Horowitz, Ashkenazy, and Goode, choosing the same trill and different performances where possible. The Fourier transform was calculated as a function of time, and the magnitude coefficients served as input for a calculation using the method of independent component analysis. In most cases this gave a more definitive determination of the onset times, as can be demonstrated graphically. For comparison identical calculations have been carried out on recordings of midi generated performances on a Yamaha Disclavier piano.
Two-voice fundamental frequency estimation
Authors: de Cheveigné Alain
Published: October 2013

An algorithm is presented that estimates the fundamental frequencies of two concurrent voices or instruments. The algorithm models each voice as a periodic function of time, and jointly estimates both periods by cancellation according to a previously proposed method [de Cheveigné and Kawahara, Speech Commun. 27, 175–185 (1999)]. The new algorithm improves on the old in several respects; it allows an unrestricted search range, effectively avoids harmonic and subharmonic errors, is more accurate (it uses two-dimensional parabolic interpolation), and is computationally less costly. It remains subject to unavoidable errors when periods are in certain simple ratios and the task is inherently ambiguous. The algorithm is evaluated on a small database including speech, singing voice, and instrumental sounds. It can be extended in several ways; to decide the number of voices, to handle amplitude variations, and to estimate more than two voices (at the expense of increased processing cost and decreased reliability). It makes no use of instrument models, learned or otherwise, although it could usefully be combined with such models. [Work supported by the Cognitique programme of the French Ministry of Research and Technology.]
Detection and representation of rhythm and tempo I: The rhythmogram
Authors: Todd Neil P. M.
Published: October 2013

In the paper an algorithm is presented for the computation of a primitive representation of rhythmic structure [N. P. Todd, J. New Mus. Res. 23, 25–70 (1994)]. The algorithm consists of the following stages: (1) cochlear filtering, by means of an ear model based on the gammatone filter-bank [M. Cooke, Modeling Auditory Processing and Organisation (Cambridge University Press, Cambridge, MA, 1992)]; (2) modulation filtering, by means of a Gaussian low-pass modulation filter-bank; (3) peak detection and integration of the output of the modulation filter-bank. When the peak outputs of the modulation filters are plotted in a time-constant/time diagram they form a representation, referred to as a rhythmogram, which resembles a more conventional tree diagram of rhythmic structure. A single acoustic event gives rise to a single branch in the representation so that integration along a branch yields a single number corresponding to the perceptual prominence or stress of the event. The integration mechanism can be thought of as an acoustic sensory memory process and accounts for a number of important phenomena in rhythm perception, including persistence, partial backward masking, interval produced accents and grouping by temporal proximity and accent. This representation may form the input to higher level learning and recognition mechanisms.
Optimally tuned vibration absorbers to control sound transmission
Authors: Grissom Michael , Belegundu Ashok , Koopmann Gary
Published: October 2013

A design optimization method is proposed for controlling broadband vibration of a structure and it concomitant acoustic radiation using multiple-tuned absorbers. A computationally efficient model of a structure is developed and coupled with a nonlinear optimization search algorithm. The eigenvectors of the original structure are used as repeated basis functions in the analysis of the structural dynamic re-analysis problem. The re-analysis time for acoustic power computations is reduced by calculating and storing modal radiation resistance matrices at discrete frequencies. The matrices are then interpolated within the optimization loop for eigenvalues that fall between stored frequencies. The method is demonstrated by applying multiple-tuned vibration absorbers to an acoustically-excited composite panel. The absorber parameters are optimized with an objective of maximizing the panel’s sound power transmission loss. It is shown that in some cases the optimal solution includes vibration absorbers that are tuned very closely in frequency, thus acting effectively as a broadband vibration absorber (BBVA). The numerical model and design optimization method are validated experimentally, and the BBVA is found to be an effective noise abatement tool.
Diffraction by jagged edge noise barriers
Authors: You Jeong Ho , Menounou Penelope
Published: October 2013

Experimental data have shown that the noise barrier’s performance can be improved by introducing jaggedness into the top edge of the barrier. In the present work a combined experimental/numerical investigation was performed. The numerical investigation employs the Directive Line Source Model [Menounou et al., J. Acoust. Soc. Am. 107, 103–111 (2000)] and reveals the basic global characteristics of a jagged profile that increases the shielding effect over a straight edge barrier having the same average height. For a jagged edge profile consisting of finite-length segments inclined with respect to a straight top edge it was found that its performance depends on the inclination angle of the finite-length segments with respect to the horizontal, the directivity of the diffracted field, and the length of the inclined segments with respect to the spatial duration of the incident sound signal. It was also found that jagged profiles offering an increased shielding effect for certain receiver locations are different from jagged profiles offering moderate improvement over many receiver locations. The numerical predictions were verified experimentally. Experiments were carried out in air using a spark source generating N-shaped pulses, a capacitor microphone and a rigid aluminum plate with various jagged top edge profiles. [Work supported by Texas ATP.]
Design and fabrication of a laboratory device for demonstration of Helmholtz resonators
Authors: Melchior Marissa , Gaspar Zoren , Naghshineh Koorosh
Published: October 2013

There is an increasing demand for noise emission reduction in many applications today. These applications include automotive exhausts, fan intakes and exhausts, and air conditioning ducts among many others. One way to attenuate the noise produced by these applications includes the use of a Helmholtz resonator. A Helmholtz resonator, also referred to as a side branch resonator to a duct, is simply a cavity with a known volume connected via a neck to a duct. A simple equation relates frequency and resonator geometry such that the length and cross section of the resonator neck, as well as the volume and shape of the resonator cavity, affect resonator performance. In practice a resonator is made to target a certain frequency of noise, which is to be suppressed. The noise reduction achieved is dependent on the accuracy of construction of the resonator. A laboratory setup was constructed which incorporated many of the most influential variables. This setup will be used as a demonstration device as well as a laboratory apparatus in the Mechanical Engineering curriculum at Western Michigan University.
Noise source identification and control of a contractor grade table saw
Authors: Bleedorn Kristin , McKee Matthew , Yarbough Dale , Yu Chen , Zechmann Edward L. , Mann J. Adin
Published: October 2013

Sponsored by the National Institute for Occupational Safety and Health (NIOSH) as part of their initiative to explore noise reduction strategies for construction equipment, a team of engineering students at Iowa State University studied a contractor grade table saw. Based on standards, published work, and preliminary tests, a repeatable noise measurement procedure was developed for the table saw operation. The wood-feed rate and force were measured. With the saw operating in a standard and consistent manner, noise sources on the saw were identified using sound intensity measurement techniques and through the application of noise control strategies to individual sources. At this stage, noise control strategies, such as enclosing the motor, are effective for noise source identification but not practical. The effectiveness of both approaches to identifying the noise sources will be discussed. Based on rank ordering the contribution of each noise source to the overall sound levels, permanent noise control strategies are suggested.
Noise control of a circular saw
Authors: Okrzesik Jeff
Published: October 2013

Construction workers are subjected to high levels of noise pollution on a daily basis. Sound levels of about 85 dB(A) with exposures of 8 hours per day will result in permanent hearing loss after many years [Noise and Hearing Loss, NIH Consens Statement Online 8(1), 1–24 (1990)]. There are numerous sources contributing to the noise problems. Tools are of particular interest considering the amount of time the average worker uses them. A circular saw is a common tool used daily by many construction workers. The levels of noise produced by a circular saw will be tested in order to evaluate the problem. The noise levels as well as the locations of the noise sources will be evaluated. Several ideas regarding the reduction of sound emissions have been researched. There will be three initial focal points of the investigation. The blade properties, including the material, size and tooth design, are all key elements that will be studied closely. The numerous dynamic parts in the motor are also large sources of noise that may be reduced by increasing the efficiency of the motor. Another vital element to reducing noise involved with a circular saw is to study the interaction noise from the piece being worked on.