bluuebunny/crossref_metadata_2025 huggingface dataset:
Compositional spatial sound manipulation: Historical overview and analyses
Authors: Begault Durand R.
Published: February 2014

Since all listening has inherently a spatial aspect, it was inevitable that composers would manipulate spatial acoustic perceptions for their listeners just as they manipulated pitch, timbre and rhythm. Approaches to organizing the spatial component of a musical composition over the evolution of western European art music might be organized in terms of the following techniques: alternation between locations (e.g, polychoral music tradition, call and response), suggestion of environmental contexts and distant sound sources (in both 19th century orchestral music and later, with development of artificial reverberation, in electronic music), and sound source movement. Although not manipulated directly by the composer, performance space acoustics also contribute spatially; data on the temporal-spatial diffusion of reverberation in large performance spaces will be presented. With the development of loudspeaker and headphone spatial sound reproduction techniques, numerous resources and immense possibilities for spatial manipulation have occurred, including the formation of virtual performance spaces. In particular, binaural (3-D sound) signal processing techniques, room modeling auralization and environmental mapping have affected the composition of sonic landscapes in many ways, and will continue to do so.
Musical considerations in the design of 3-D-sound rendering software
Authors: Kendall Gary , Mann David , Robbin Scott , Kendall Alan
Published: February 2014

The rapid growth of 3-D-sound technology has created a need for 3-D audio tools which can be used by musicians and composers in much the same manner that visual artists use 3-D graphics tools. Experience with such audio tools reveals some issues of special concern for music. For example, the rapid movement of musical sound sources can create Doppler shifts that produce harsh detunings of pitch, and the realistic rendering of intensity loss with changing distance can cause some musical elements to be buried in the mix. Consider too that individual instruments are best spatialized in different environments—high strings in large reverberant halls, electric basses in small dry rooms—and that some musically useful spatial effects, like stereo decorrelation, are not conveniently produced with environmental models. Spatial sound rendering software needs to provide numerous exceptions to accurate physical modeling in order to adapt to the musical context and it must support a heterogeneous collection of musically useful tools. In fact, the true goal in the design of 3D software for music should be to provide a creative tool that captures the complexity and richness of spatial sound without slavishly adhering to models of acoustic reality.
Model-based virtual room acoustics
Authors: Thompson Charles , Dennis Max , Tsui Jing , Raspopvic Miroslava
Published: February 2014

A new procedure for evaluating the impulse response between source and receiver locations in small reverberant rooms is outlined. The modified image method uses the solution of the Laplace transform based formulation for the Sommerfeld integral to determine the strength of image sources and boundary reflections. The model is used to simulate sound spatialization effects in small rooms. It is shown that spatial invariance of the poles in the room transfer function allows one to use IIR models for the long-time behavior of the impulse response. The variability of the short-time response with changing observation location is to be modeled using FIR models.
Directionless surround mixing
Authors: Colburn Scott
Published: February 2014

The first commissioned audio DVD, titled Immersions, contained a piece by Ellen Fullman and her long stringed instrument. The concept of mixing this piece for surround sound was complicated because it could not be rehearsed. It was conceptualized in advance without hearing it. It is tempting to use surround sound (like quad) to fly sounds around in the surround field but this novelty will wear thin after repeated listening. This mix was created to utilize movement within the surround field through the use of phantom images. The mix also has a defined front and back, but remains interesting wherever you sit in the sound field regardless of how you face. It also had to be good in stereo.
Assessing the effectiveness of national and international noise policies and regulations
Authors: Finegold Lawrence S. , Schomer Paul D.
Published: February 2014

This I-INCE technical initiative will assess the effectiveness of noise policies and regulations around the world. During the last half of the 20th century, many countries have recognized noise as an environmental and occupational problem, and various countries have been working to develop adequate noise exposure policies and noise control technologies. Considerable time and effort are devoted each year to develop noise exposure policies for places where people work, where people live, and for outdoor environments devoted to leisure activities. However, very little is known about how effective these noise policies and regulations have been in controlling the noise exposure of the populations which they are intended to protect. This policy research initiative involves a study of existing noise exposure policies and regulations in countries which have recognized noise as a problem affecting the public health and welfare. The first phase involves the collection and cataloging of noise exposure policy guidelines, regulations, and standards. The second phase involves developing a baseline of noise exposure estimates for each participating country. The third phase will entail determination of the long-term effectiveness of these policies and regulations in controlling noise exposure by examining the changes in various population noise exposure parameters over time.
Status of the I-INCE technical initiative on noise from outdoor recreational activities
Authors: Miller Nicholas P.
Published: February 2014

Outdoor recreational activities are increasingly both creating noise and being affected by noise. A study group has been formed and met for the first time at Internoise 2000 in France. The Convener of the study group is Dr. Philip Dickinson from New Zealand, and there are approximately 12 other members who represent various countries around the world. The group is focusing both on the effects on the environment (people as well as wildlife and natural habitats) resulting from noise produced by outdoor leisure activities, as well as on the effects on leisure activities of noise from other activities. The study will consider national parks and wilderness areas, will examine the need to preserve general quiet areas, and will identify the effects on leisure activities of various sources of noise. Members have been gathering information, and the study group expects to identify specific recreational activities, with each member focusing on a particular activity, according to his or her expertise. In this presentation, the author will identify some of the specific technical and policy issues that can arise when developing a quantitative approach to monitoring and analyzing sound environments in park and wilderness areas.
Growth of masker level with signal level as a function of preceding stimulation
Authors: Strickland Elizabeth A.
Published: February 2014

Some recent data suggest that stimulating the auditory system with sound may turn down active processing. The basic paradigm used to show this is to measure the level of a notched-noise, long-duration masker required to just mask a short-duration tone that occurs near masker onset. The preceding stimulation has been a precursor that was either a fixed-level broadband noise [Strickland, J. Acoust. Soc. Am. (in press)], or an extension of the notched-noise masker, which varied in level [Strickland, J. Acoust. Soc. Am. 107, 2915 (2000)]. The present experiment was done to examine the effects of a notched-noise precursor that had the same frequency characteristics as the masker, but which would be fixed in level. The signal was a 10-ms, 4-kHz tone. The notched-noise masker was 200 ms, and the precursor was 205 ms. For a given precursor level, the growth of masker level with signal level was determined. This data was used to calculate the effective input–output functions of the signal and the masker. These functions will be compared to those published physiological measures from the chinchilla cochlea. [Work supported by NIH (NIDCD).]
Do listeners include the echo portion of a damped sound when judging its duration?
Authors: Schlauch Robert S. , DiGiovanni Jeffrey J. , Olson Kristi L. , Donlin Erin E.
Published: February 2014

Ramped sounds (slow attack, fast decay) are perceived to be longer in duration than damped sounds (fast attack, slow decay) that are the same physical duration even when they are simply time-reversed versions of each other. Stecker and Hafter [J. Acoust. Soc. Am. 107, 3358–3368 (2000)] suggest that this perceptual difference is a result of listeners ignoring a portion of the decay of damped sounds because they consider it an ‘‘echo.’’ To test this idea, a duration-matching experiment with two groups of 10 listeners was completed. Each group matched the duration of a broadband noise to two natural sounds, a drum strike and a word spoken in a reverberant room, presented in the normal fashion and reversed in time. The first group was told to match the duration of the sounds. The second group was told to consider the entire duration of the sound, even portions that sound like an echo. The ratio of ramped/damped at equal subjective duration was larger for the first group than for the second group. This finding supports Stecker and Hafter’s idea that listeners ignore the echo portion of the sound, but it did not account for the entire effect. [Work supported by NIH.]
Spectral modulation transfer functions as a function of duration and level
Authors: Eddins David A. , Harwell Ross
Published: February 2014

Spectral modulation transfer functions (SMTF) were estimated as a function of stimulus duration and level to evaluate the influence of those parameters on spectral shape perception. Listeners discriminated between noise stimuli having either a flat or a sinusoidal spectral shape. Spectral contrast thresholds were determined by adaptively varying the peak-to-valley contrast using a 3-down, 1-up tracking algorithm in a cued, two-interval, forced-choice procedure. Carrier noises had bandwidths of either 6 octaves (200–12800 Hz), or 1 octave (800–1600 Hz, 6400–12800 Hz). Spectral frequencies ranged from 0.25 to 10 cyc/oct and 1.33 to 10 cyc/oct for the 6- and 1-octave conditions, respectively. In Experiment I, the stimulus durations were either 50, 100, 200, 300, or 400 ms. SMTFs were bandpass in shape with best contrast thresholds between 2 and 5 cyc/oct. Decreasing duration had a greater influence on thresholds for low- and high-spectral frequencies than mid-spectral frequencies. In Experiment II, stimulus level was varied in spectral levels from –10 to 50 dB SPL. The shape of the SMTF varied little with level down to No = 10 dB. Decreasing level had a greater influence on low- and high-spectral frequencies than mid-spectral frequencies. [Work supported by NIH NIDCD RO1DC04403.]
Sample discrimination of target-tone duration with variable-duration context tones
Authors: Neff Donna L. , Reimer Jason F.
Published: February 2014

This study examined the ability of three normal-hearing listeners to discriminate duration differences in target tones with and without variable-duration context tones in frequency regions remote from the target. Single target tones at 500 (‘‘low’’), 1260 (‘‘middle’’), or 3176 Hz (‘‘high’’) were combined with one or two context tones in nontarget regions (e.g., a low target paired with a high context). Across intervals in the 2AFC task, the target tone was drawn from Gaussian distributions having a mean duration of 100 or 120 ms. Listeners were to select the target drawn from the distribution with the longer mean duration. Context-tone duration was sampled from this ‘‘correct’’ distribution, so duration alone did not cue the targets. Overall performance and perceptual weights for individual tones were examined. Frequency had no effect on performance for targets alone, however, there were interactive effects of target and context frequency. The worst performance was observed for middle targets with flanking context tones. Flanking context tones were effective in combination, but relatively ineffective individually. For low and high targets, the context tone nearer the target frequency was most effective, but both influenced performance. Comparisons to similar conditions with frequency sample discrimination will be discussed. [Work supported by NIH.]
Investigating the nature of cross-language perceptual comparisons: Evidence from production
Authors: Baker Wendy , Trofimovich Pavel
Published: February 2014

In cross-language speech perception research, a metric of perceived phonetic similarity between native and non-native sounds both predicts and describes non-native listeners’ perceptual accuracy [Guion et al., J. Acoust. Soc. Am. 107, 2711–2724 (2000)]. By hypothesis, judgments of perceived phonetic similarity reflect the match between a non-native sound and the sound category used to process it. Whereas measures of perceptual accuracy often mask the nature of the perceptual categories used to process non-native sounds, the acoustics of non-native segment articulations may unambiguously indicate whether non-native sounds are processed in terms of perceptually similar native or newly established non-native sound categories. This study examines the relationship between judgments of phonetic similarity between Korean sounds and six English vowels in bVd and bVt words and productions of the same vowels by 40 Korean listeners who differed in age and amount of exposure to English. A metric of perceived phonetic similarity between English and Korean vowels was correlated with results of acoustic comparisons between English stimulus tokens and the listeners’ productions, revealing that a metric of perceived phonetic similarity also predicts production accuracy and that the nature of the sound categories used to process non-native sounds determines cross-language perceptual relationships.
Perception and production development in citation versus sentence contexts by learners of Japanese
Authors: Hirata Yukari
Published: February 2014

This study compares the degrees of L2 learners perception versus production development in isolated word contexts with those in sentence contexts. Native speakers of English learning Japanese participated in the perceptual training program used in Hirata [Proc. Int. Congr. Phon. Sci. 2, 1413–1416 (1999)] in which subjects identified, and received immediate feedback on, the numbers of morae in target words in either citation or sentence contexts. Before and after training, subjects were tested on their ability to accurately perceive and produce words that included difficult durational contrasts (1) in citation contexts (word tests), and (2) in frame sentences (sentence tests). The results of the word tests showed no positive or negative correlation between the degrees of perception and production improvement. However, in the sentence tests, a negative correlation was found (r=−0.651): the greater the improvement in one domain, the less the improvement in the other. These contrasting results suggest that development captured in isolated word contexts does not readily predict that in fluent speech contexts, expanding Greenspan et al. [J. Exp. Psychol. 14(3), 421–433 (1988)]. The factors responsible for the negative correlation in the sentence contexts are discussed in light of the mechanisim of fluent L2 acquisition. [Work supported by Consortium for Language Teaching and Learning.]
Speaking clearly for non-native listeners
Authors: Bradlow Ann R. , Bent Tessa , Schwarzlose Rebecca
Published: February 2014

Non-native listeners often have particular difficulty perceiving novel phoneme contrasts, and are disproportionately affected by background noise relative to native listeners [e.g., Mayo et al., J. Speech Hear. Res. 40, 686–693 (1997); Meador et al., Bilingualism 3, 55–67 (2000)]. This study investigated whether these difficulties could be overcome by naturally produced clear speech. Based on previous findings that hearing-impaired adults [Picheney et al., J. Speech Hear. Res. 28, 96–103 (1985)] and learning-impaired children [Bradlow et al., J. Acoust. Soc. Am. 108, 2603 (2000)] derive great benefit from clear speech (17%–20% and ∼9%, respectively), we predicted that non-native listeners would also exhibit a substantial clear speech effect. The 32 non-native and 32 native listeners were presented with English sentences. Factors that varied were speaking mode (conversational versus clear), signal-to-noise ratio (−4 vs −8 dB) and talker (male versus female). Contrary to our prediction, non-native listeners exhibited a relatively small clear speech effect (∼5%). This result contrasts sharply with the large clear speech effect for hearing-impaired adults and learning-impaired children. We interpret these cross-population differences as reflecting differential effects of speech perception deficits that arise from problems of access to the underlying linguistic code versus those that arise from problems of access to the signal.
Perceptual weighting in fricative/vowel syllables for speakers of English and Mandarin
Authors: Radhakrishnan Sreedivya , Schmidt Anna Marie
Published: February 2014

Adult English speakers have been shown to pay more attention to noise spectra in labeling syllable–initial fricatives than to formant transitions. However, cross-language research indicates that languages differ in weights assigned to aspects of the acoustic signal. In this study, native speakers of English and Mandarin Chinese labeled and rated for goodness (in the L1) both English and Mandarin alveolar and post-alveolar fricative noises from four vowel environments. They also labeled and rated natural FV syllables in each language. Finally, subjects heard F + V combinations (from their L1 only) in which fricative noise from the anterior fricative was combined with vowel transitions from more posterior fricative and vice versa. Vowel length was manipulated in one set of stimuli. Results indicate differences in weighting by the two groups of subjects.
Acoustic study of vowel nasalization in Thai
Authors: Onsuwan Chutamanee , Speeter Beddor Patrice
Published: February 2014

Patterns of coarticulatory nasalization and segmental duration were investigated in Thai, a language with contrastive vowel length. Four Thai speakers produced multiple repetitions of real and nonsense CV(:)N items, where C=[b], V=/i e ae/, V:=/i: e: ae:/, and N=bilabial, alveolar, or velar nasal. Three temporal measures were taken: duration of V(:), N, and vowel nasalization. Acoustic onset of vowel nasalization was primarily identified from FFT spectra, sampled across the vowel in 10-ms increments, as onset of the nasal formant and/or broadening of formant bandwidth in the low-frequency region. Preliminary results from two speakers showed that long V: was nearly twice as long as short V, and final N was nearly two times longer following short V than long V: [A. S. Abramson, Int. J. Am. Linguist. 28 (1962)]. Vowel nasalization varied as a function of V(:)N duration: relative to N onset, nasalization started earlier in long V: than short V. However, long V: was proportionately less nasalized (44% for long V: vs 49% for short V). These results are compared to findings for English, and interpreted relative to look-ahead and coproduction models of coarticulation.
Multipath of sound propagation in shallow water and underwater spread spectrum communication
Authors: Chen Yan , Chen Geng
Published: February 2014

Due to the influence of the sea bottom and the sound speed profile, the propagation of sound in shallow water appears to be a complicated multipath. The time spread and the amplitude fluctuation of different arriving paths cause strong fading and intersymbol interference (ISI). At the receiving point the multipath signal can be expressed as r(t)=A0s(t−τ0)+∑i=1N-1Ais(t−τi)+n(t), where the first term is arriving signal directly, the second term is arriving signals by different paths, the third term is noise. When the difference of time delays is over the duration of code chip, a strong ISI occurs. As we know, the pseudo-random signal has high resolution in both time and frequency domains. It is often used in spread spectrum communication. In this paper a diversity technique with spread spectrum is used for realizing coherent accumulation of the arriving paths. The results of the lake experiment and the sea experiment are shown in this paper. The efficiency of coherent accumulation is obvious.
Speech privacy in open-plan offices. II. Developing a model
Authors: Wang Chong , Bradley John
Published: February 2014

This work gives an overview of the development of a mathematical model of speech propagation between workstations in a conventional open-plan office. The work was carried out in two stages. A model was first developed for sound propagation over a single screen and including realistic floor and ceiling reflections. Maekawa’s screen diffraction result was used and reflections were modeled using an image sources technique. Interference effects are seen to be important at lower frequencies but a simple energy addition of diffracted and reflected sound gives adequate results at speech frequencies. The second phase of the model also included reflections due to the panels of a complete pair of rectangular workstations and again using an image sources technique. Correctly modeling the ceiling absorption and the influence of ceiling-mounted light fixtures are seen to be critical to achieving accurate predictions. The current model was found to predict Speech Intelligibility Index values with an rms error of only 0.02.
A study on speech privacy in closed offices
Authors: Caswell Daryl J. , Ma Yong , Goodchild Jim T. , Dai Liming
Published: February 2014

Quantification of speech privacy in closed spaces is dependent on several factors: attenuation and absorption characteristics of the construction, background sound levels, speaking levels of the talker, the language and dialect, as well as accent used in the speech. Speech privacy is the opposite concept of speech intelligibility and can be assessed by the predictors of speech intelligibility. In this paper, several assessment methods of speech intelligibility are introduced. They are adapted to measure the speech privacy in closed offices, including standard ASTM E1130, speech transmission index (STI) and early-to-late ratio (clarity). The experimental results are used to evaluate the speech privacy index (PI), which is the rating number proposed for assessing the speech privacy. The subjective measurements offered by human subjects are also conducted using different languages: English and Mandarin Chinese. The test materials used in the speech intelligibility testing are single words (MRT words), sentences and conversations. Both measurements are described under the same acoustical environment. The results of the subjective and objective measurements indicate that the current standard needs some modifications when it is used in closed offices.
Brad Sturtevant’s contributions to research on shock waves and lithotripsy
Authors: Blackstock David T.
Published: February 2014

Bradford Sturtevant (1933–2000) spent his entire academic career in the Graduate Aeronautical Labs at Caltech (GALCIT). He was a distinguished and internationally respected member of the fluid mechanics community, known particularly for his research on shock waves. Although a member of the Acoustical Society since 1974, he rarely if ever published in JASA. Nevertheless, his contributions to acoustics, mainly in what we think of as nonlinear acoustics although he probably saw them simply as part of fluid mechanics, are significant. Examples are: shock focusing, nonlinear resonance in closed- and open-end tubes, sonic booms, and propagation of weak shocks through a random medium. In the late 1980s he became interested in lithotripsy. When Andrew Evan (Indiana University School of Medicine) organized a multidisciplinary consortium to solve many of the mysteries of lithotripsy, it is not surprising that Brad was picked for his expertise on shock waves and his interest in shock waves in the body. The research effort that ensued, now in its eighth year, owes much to the work of Brad and his students and to his counsel. Brad was especially concerned about the mechanism(s) by which stone breakup occurs.
Perception of single octave-related complexes in relation to the tritone paradox
Authors: Ragozzine Frank
Published: February 2014

In the tritone paradox, listeners are presented with a sequential pair of octave-related complexes separated by an interval of half an octave. Listeners tend to perceive these stimuli such that half of the pitch class circle is heard as higher, and the opposite half is heard as lower. It is theorized that listeners compare these tones to a circular pitch class template, having an orientation with respect to height that differs among listeners, in making these relative height judgments. In the present experiment, listeners were presented with two sets of stimuli. In the first, listeners were presented with the tritone paradox stimuli. In the second, listeners were presented with a single octave-related complex (rather than a pair of such tones) on each trial, and were asked to determine whether each tone was high or low without comparison to any other tone. An influence of pitch class on perceived height was shown for both tritone and single tone stimuli, and a significant relationship between perception of the tritone paradox and perception of the single octave-related complexes was obtained. These results are in accord with the pitch class template theory.
Tuning the lower resonances of carved Baltic psaltery by adjusting area of the soundholes
Authors: Peekna Andres , Rossing Thomas D.
Published: February 2014

A 12-string kannel was constructed by Ain Haas, initially without any soundholes. Subsequently, an initial soundhole pattern was drilled and then enlarged incrementally. The frequencies of the two lowest dipole modes changed significantly, from 382 and 492 Hz to 411 and 454 Hz, respectively. Introducing distributed soundholes appears to affect the internal air motion. The mode damping was also observed to increase as soundholes were added, although damping of the lowest symmetric mode decreased upon adding holes of larger diameter. The sound quality was judged to improve as more soundholes were added.
Sound localization by interaural time differences at high frequencies
Authors: Constan Zachary A. , Hartmann William M.
Published: February 2014

It is possible for human listeners to lateralize high-frequency noise on the basis of interaural time differences in the fine structure of the envelope. However, the ability to do so makes great demands on the interaural coherence of the noise. This paper explores listeners’ ability to lateralize a broadband, high-passed, coherent-noise signal in the presence of a broadband incoherent masker. Results showed that as the high-pass cutoff frequency increased through the critical region from 1 to 4 kHz, the required interaural coherence increased rapidly, especially for the smaller changes in lateral position. The observed functional behavior can be successfully predicted by a neural model of lateralization based on the centroids of bandwise cross-correlation functions of model peripheral inputs that have been rectified and low-pass filtered [e.g., Bernstein and Trahiotis, J. Acoust. Soc. Am. 100, 1754–1763 (1996)]. The results have implications for the localization of broadband sounds in rooms, where the interaural coherence tends to increase with increasing frequency, but often not rapidly enough to allow the envelope timing information to contribute usefully. [Work supported by NIDCD.]
Effect of reverberation on spatial unmasking for nearby speech sources
Authors: Shinn-Cunningham Barbara G. , Mraz Lisa , Kopčo Norbert
Published: February 2014

Individualized HRTFs were measured in a moderately reverberant room (T60=450 ms) for sources directly in front of and to the right of a listener for both near (15 cm) and far (1 m) distances. The full HRTFs (including reverberation) and pseudo-anechoic HRTFs (time windowing out the reverberation) were used to simulate a speech target and a speech-shaped noise masker over headphones. Speech reception thresholds were measured adaptively, varying the target level while keeping the masker level constant at the better ear. Thresholds were measured for both left and right monaural signals as well as for binaural signals. Results show the magnitude of spatial unmasking that can arise for sources very close to the head, where large interaural level differences (ILDs) arise, and determine the degree to which spatial unmasking is due to better ear and binaural effects. These results are compared to previous anechoic results simulating sources near a listener in which large ILDs appear to degrade binaural performance below predicted better-ear performance. Comparisons between pseudo-anechoic and realistic reverberation conditions address the degree to which reverberation interferes with spatial unmasking. [Work supported by a grant from the Air Force Office of Scientific Research.]
Measurements of the directional performance of commercial hearing aids
Authors: Schulein Robert B. , Christensen Laurel A. , Haapapuro Andrew J.
Published: February 2014

Various estimates of directional microphone performance are possible, yielding a wide variety of published and advertised signal-to-noise improvements. There appears to be a growing concensus that the ratio of on-axis sound to diffuse sound (the ‘‘directivity index’’) provides the most realistic measure for real-world use of hearing aids. We undertook to measure several commercial digital and analog hearing aids, mostly of recent design, using the methods under consideration for ANSI standard adoption: anechoic polar measurements with data numerically integrated to obtain a diffuse-field directivity index estimate and direct anechoic and reverberation-room measurements. These measurements generally agreed, and showed a wide range of performance across designs.
Structural acoustics with mean flow: Instability waves on force excited submerged elastic structures
Authors: Arzoumanian Sevag H.
Published: February 2014

Computational verification of the Crighton and Oswell predictions for the response of a line excited, doubly infinite, submerged plate in mean flow are presented. The phenomena highlighted for validation include: the occurrence of absolute instability above a critical flow speed, the presence of convectively unstable waves downstream from the driver for all flow velocities below this critical speed, and the existence of anomalous propagating waves with group velocities pointed towards the driver over a narrow range of excitation frequencies. Extensions of the Crighton and Oswell model to include the effects of compressibility in the fluid, finite thickness in the structure and the presence of compliant coatings are discussed. Extensions of this model to three dimensions are also presented through analytical studies and numerical simulations of instability waves on a point excited 2-D plate submerged in a 3-D fluid in mean flow. [Work supported by ONR and St. John College, Cambridge.]
Speech enhancement for an in-ear communication system using bone-conducted and acoustic signals
Authors: Papanagiotou Kyriakos , Rafaely Boaz , Shadle Christine H.
Published: February 2014

The main requirement for mobile radiotelephony nowadays is an ergonomic, robust, and functional system. One weak point of existing devices, such as hands-free kits for mobile telephones, is poor speech intelligibility due to factors such as environmental noise and poor signal reception. This work investigates the potential of speech enhancement using post-processing of signals from two types of transducers: a miniature microphone placed on the opening of the ear canal, and a miniature accelerometer mounted on four locations around the head. The microphone provides good speech quality with the drawback of being sensitive to external noise, while the accelerometer senses vibrations that are immune to ambient noise, but are a distorted, low-pass-filtered version of speech sounds. The use of optimal filtering, which enhances the signals acquired from the miniature transducers by matching them to a reference recording of the same material, was investigated. The methods investigated here could then be applied to improve speech quality and reduce noise in communication earsets. Initial results with speech, based on sustained phonemes and phonetically balanced sentences, showed potential for improvement in the quality of the speech signals and also suggested that different filters might be used for different phonemic contexts. [Research funded by EPSRC Grant No. GR/M86026.]
Laser Doppler vibrometer and subminiature electret microphone: New methods for objective evaluation of nasality
Authors: Kumada Masanobu , Kaneko Toshiaki , Ono Takahiko , Ohashi Masanao , Ishida Koji , Niimi Seiji
Published: February 2014

It is very important to evaluate nasality objectively in terms of both intersubject (severity of nasality) and intrasubject (validity of treatments, i.e., operation, speech therapy etc.). But there is no definite methodology for an objective evaluation of nasality. Our objectives were to evaluate the degree of nasality with a laser Doppler vibrometer (LDV) and a subminiature electret microphone (SEM) (using an accelerometer as a known standard), and to discuss the possibility of clinical use of these methodologies. LDV, used recently in the industrial field as well as in the medical/biological field, is a generator and detector of lasers to detect the velocity of the object’s vibration by the Doppler effect. The subjects were five healthy Japanese males (Tokyo dialect speakers). Used were sustained phonations (about 2 s) of /m/ and five Japanese vowels (/i/, /e/, /a/, /o/, /u/), and the Japanese words, shinkansen and shimbunshi (six repetitions for each phoneme). By both LDV and SEM systems, a significant difference was found beween /m/ and each vowel, which means that both systems can detect nasal/non-nasal differences and seem valid to evaluate the severity of pathological nasality (e.g., a cleft palate). Further study is planned for not only sustained phonation and word but also for sentence level, using patients (pathology) as well as healthy subjects (physiology). Furthermore, LDV seems to be a powerful tool for speech physiology where vibration of the body wall can be measured during speech and singing, without any contact receiver.
Acoustic-phonetic approach to speech recognition based on event detection and linear discriminant analysis
Authors: Juneja Amit , Espy-Wilson Carol
Published: February 2014

An approach to speech recognition based on phonetic-feature theory was developed to recognize the E-set. The speech signal is first segmented into broad classes based on parameters for the manner phonetic features sonorant, syllabic, continuant, and strident, in addition to silence. The speech signal is divided into the regions: vowel, sonorant consonant, strong fricative, weak fricative, and stop. The events specified by the manner features are then used to extract parameters relevant for the place phonetic features. In particular, we extract parameters for the place phonetic features labial and alveolar for stops; and alveolar and palatal for strong fricatives. Decisions at this stage of recognition are based on an optimal linear combination of parameters. The linear weights are obtained from the training data by applying linear discriminant analysis. The TI46 database is used for training and testing. The results of our event-based system are compared to a standard MFCC and HMM based recognizer. [Work supported by NSF Grant SBR-9729688.]
Robust speech event detection using strictly temporal information
Authors: Deshmukh Om D. , Wilson Carol E. , Salomon Ariel
Published: February 2014

A major problem in the development of robust speech recognition systems is our understanding of how to deal with noise and/or reduced spectral information. Previous studies have shown that the temporal structure of speech is robust and a good source of information for recognizing manner. This work discusses the development of algorithms that specifically target temporal information in the speech signal that can be used to identify different manner classes. In particular, we extract events associated with sonorant regions and obstruent regions. We compared the performance of our algorithms on clean speech (sentences taken from the TIMIT database) and on 1 channel spectrally impoverished speech [as described by Q.-J. Fu, House Ear Institute, Dept. of Auditory Implants and Perception, Tigersoft]. The results show no degradation in performance for the detection of landmarks associated with stops, affricates, strident fricatives, and vowels. There is a small degradation of six landmarks associated with the ‘‘weaker’’ consonants (nonstrident fricatives and sonorant consonants).
Robustness of beamforming algorithms with head-related transfer functions
Authors: Lockwood Michael E. , Jones Douglas L. , Bilger Robert C. , Lansing Charissa R. , O’Brien William D. , Wheeler Bruce C. , Feng Albert S.
Published: February 2014

This study examines the robustness of the frequency-banded minimum-variance beamformer (FBMVB) when processing signals from in-the-ear microphones or signals that include head-related transfer function (HRTF) effects. In previous work using simulated signals from a free-field array [Lockwood et al., J. Acoust. Soc. Am. 106, 2278 (1999)], the two-channel FBMVB proved to be highly effective in extracting the signal of a target sound from a known direction amidst multiple interfering sound sources from other unknown directions. When the signals were modified to include the effects of a measured HRTF [G. Gardner and K. D. Martin, J. Acoust. Soc. Am. 97, 3907–3908 (1995)], the performance of the FBMVB improved as measured by the SNR gain and intelligibility-weighted SNR gain. The FBMVB also outperformed conventional algorithms in both the free-field and HRTF tests. Additionally, analytical results show that under certain conditions the FBMVB may extract a target source and perfectly cancel multiple interfering sound sources despite differences in the frequency-dependent intersensor gains for each source. Consequently, the FBMVB may extract signals from the on-axis direction without knowledge of the HRTF, as long as the HRTF in the on-axis direction is identical for both ears. [Work supported by NIH-NIDCD Grant R21DC04840.]
Shallow-water reverberation from a time reversed mirror: Data-model comparison
Authors: McDonald B. Edward , Holland Charles
Published: February 2014

A July 1999 bottom reverberation experiment and accompanying simulation results will be discussed. The experiment was carried out as part of the Focused Acoustic Fields joint program between Saclantcen and Scripps Institution of Oceanography. A 3.5-kHz probe source 1 m off the bottom just north of Elba produced a signal which was recorded on a vertical source/ receive array (SRA) at ranges between 3 and 7.4 km. The SRA then time reversed and retransmitted the signal, resulting in a focal annulus near the bottom at the probe source range. Reverberation from the SRA signal was recorded at the SRA. A distinct maximum in reverberation at the time corresponding to the probe source range indicates selective bottom ensonification near the focus. Simulation studies carried out using RAM and the Saclantcen PROSIM reverberation models reproduced salient features of the experiment and led to suggestions for future experiments. Model sensitivity to array aperture and focal range will be discussed. Results imply that reverberation from a time reversed mirror may be useful in rapidly probing mildly varying bathymetry. [Work supported by Saclantcen.]
Echo to reverberation enhancement with time-reversal methods in the ocean
Authors: Kim Seongil , Kuperman William A. , Hodgkiss William S. , Akal Tuncay , Song Hee Chun , Edelmann Geoffrey F.
Published: February 2014

Reverberation from a rough ocean boundary often degrades the performance of active systems in shallow water. In this study, two time-reversal methods are investigated to enhance the echo-to-reverberation ratio by focusing more acoustic energy in the water column rather than on the ocean boundary. The first method is with a probe source in the middle of the water column. A time-reversal process with the probe signal realizes a focus at the probe signal position and shadows the boundaries below and above the focus resulting in reduced reverberation and increased target echo. The second method is without a probe source. An eigen-decomposition of the reverberation signal provides the transfer function information between the source and the scatterers where the reverberation is generated. The backprogation from a source array weighted by nondominant eigenvectors minimize the acoustic field incident on the corresponding boundary. Both of these methods place a notch in the returning reverberation signal. The results from numerical simulations as well as an ocean experiment will be discussed. [Work supported by ONR.]
The meaning of negative group velocity and amplitude modulated signals
Authors: Werby M. F. , Uberall H.
Published: February 2014

Negative group velocities have been reported in calculations for waves generated on elastic objects. It is usual to interpret a group velocity with the travel time of a wave front or the time that the energy of a wave generated travels along some trajectory. One may look at the group velocity as the rate at which an envelope of a wave train travels. More generally, one may exploit the principle of stationary phase along with the Fourier transform in frequency and derive the concept of group velocity from the first order asymptotic term. This derivation requires that the modulus of the integral is changing slowly relative to the phase term. In addition it is required that the group velocity is not near a stationary point in frequency. In that case first order asymptotics is no longer valid and one must employ higher asymptotics. Here, the standard concept of group velocity is no longer meaningful. Further, the signal becomes amplitude modulated rather than frequency modulated. We examine this for some examples, assuming losslessness. In the event of loss the concept of group velocity has already been reported to be of dubious value.
Solution of the nonhomogeneous Sturm–Liouville problem and an inverse solution to an inclusion imbedded in a waveguide
Authors: Werby M. F. , Uberall H.
Published: February 2014

The homogeneous Sturm–Liouville problem is the usual starting point of a normal mode theory provided one adds absorption as a perturbation after the nonlossy solution in a waveguide. In this development we assume that prior to an inclusion the solution is known for the waveguide problem for any given source. When an inhomogeneity is introduced into the waveguide we assume that it may be treated as a source term. Using completeness, we expand the source term as a sum of eigenfunctions with unknown coefficients. Under reasonable conditions measurements of the field of the waveguide before and after the presence of the inhomogeneity allows one to determine the expansion coefficient of the source term and thus many relevant properties of the inclusion itself. Thus, determination of the source term allows one to determine features of an inclusion.
Acoustical design of baseball stadiums
Authors: Marsh David E.
Published: February 2014

Baseball team owners and stadium managers alike wish to have good acoustics in their venues. To achieve this, many issues must be addressed early in the design of a new stadium. For example, if community noise is a concern for a new open-air stadium, addressing this will require construction budgeting for certain architectural and sound system design features. Minimizing echoes is an acoustical design challenge for both open-air and enclosed stadiums. This requires a general awareness of likely echo paths, knowing what acoustical treatment options are generally acceptable for this building type, and use of predictive modeling for the sound system design. Enclosed stadiums pose additional design challenges such as the quieting of very large air-handing units, minimizing reverberation, and making the sound system intelligible. These challenges extend beyond the seating bowl to the concourses and a host of ancillary areas. This paper reviews the authors’ experience in the overall acoustical design of modern baseball stadiums.
Effect of a rigid boundary on bubble dynamics in ESWL
Authors: Ohl Claus-Dieter , Geisler Reinhard , Lauterborn Werner , Prosperetti Andrea
Published: February 2014

Recently it has been observed that two countertraveling lithotripter pulses form a cavitation bubble cloud with band-like structure of bubble activity [D. Sokolov et al., J. Acoust. Soc. Am. 107, 2838 (2000)]. A similar structure becomes apparent after focusing a single lithotripter pulse onto a rigid boundary. Here, the growing cavitation bubbles interact with the reflected pulse and the band-like structure appears near the boundary. Experiments will be presented to compare the nucleation, explosive growth and shock wave interaction of cavitation bubbles near the reflector to bubbles in a free liquid. The experiments have been conducted by means of high speed photography (up to 10 million frames per second) employing microscopic imaging. The observed radial and translational motion will be compared with a force balance model, taking into account the pressure radiation force.
Bubble dynamics in a lithotripter shock wave
Authors: Storey Brian , Szeri Andrew , Matula Tom
Published: February 2014

Typically, a pulse in shock wave lithotripsy has an intense shock front followed by a long negative pressure tail. A micron sized bubble subjected to this pulse is compressed rapidly by the shock, before growing to millimeter size in response to the negative pressure tail. As this tail passes, the bubble will then undergo an essentially free collapse and subsequent after-bounces. Calculations of the bubble dynamics using a model that takes phase change, heat transfer, mass transfer, and chemical reactions into account provides excellent agreement with experimental measurements. The role of phase change and mass transfer is dominant and the bubble is composed primarily of vapor throughout the free collapse and after-bounces. When the vapor bubble collapses there are significant chemical reactions and predictions on the amount of radical species produced and peak temperature achieved will be presented.