bluuebunny/crossref_metadata_2025 huggingface dataset:
Nonzero threshold loudness: A circular argument in Zwislocki (1965) and Moore, Glasberg, and Baer (1997).
Authors: Nizami Lance
Published: October 2013

ANSI S3.4-2005 updates the calculation of loudness to reflect empirical evidence that loudness is nonzero at detection threshold. Hellman [Acoustics Today (2007)], in reviewing the changes, cited the work of Zwislocki [Handbook of Mathematical Psychology (1965) Vol. III] and Moore et al. [J. Audio. Eng. Soc. 45 (1997)] as theoretical support. Zwislocki proposed that detection threshold reflects an internal noise that acts like an external masker, and that (using generalized notation here for clarity) the tone+noise loudness for an rms tone pressure x is a function f of the sum of noise contribution(s) c and a tone contribution g(x). The Zwislocki g(x) was zero at x=0 and increased monotonically thereafter. Altogether, tone+noise loudness is f(g(x)+c). Zwislocki then assumed that listeners can perceptually separate tone from noise. He subtracted noise loudness from tone+noise loudness to get tone loudness, [f(g(x)+c)−f(c)]. That exceeds zero for whatever x is deemed the tone-detection threshold. Unfortunately, with g(x)>0 for x>0, a nonzero threshold-tone-loudness was predetermined. Moore et al., using auditory filter output power as x, produced a congruent equation for tone loudness in quiet, repeating the circularity. Circularity was inevitable, because detection threshold is defined using a percentage-correct performance that indicates nonconstant loudness.
Tone-glide direction identification revisited.
Authors: Feth Lawrence , Hoglund Evelyn
Published: October 2013

Dawson and Feth (2004) reported that listeners showed an apparent difference in sensitivity for the direction of frequency change when listening to virtual frequency glides or frequency modulated (FM) tones. Their work was a partial replication and extension of the work by Gordon and Poeppel (2002) using a one-interval fixed-block paradigm. In each interval, the listener was asked to report whether the frequency was rising or falling using buttons labeled up and down. Psychometric functions were reported but the original data were not examined for interval response bias. This study addresses possible bias in two ways. The original data were reexamined for response bias, and direction identification was tested using FM glides in the single-interval adjustment-matrix (SIAM) procedure. The SIAM procedure [Kaernbach, (1990)] is designed to control response bias in a one-interval paradigm. Contrary to the previous work, data collected with the SIAM procedure do not show a consistent pattern of sensitivity toward change in either direction. Additionally, psychometric functions using this procedure are steeper than those of the prior studies. Implications of these results in comparison with previous work will be discussed. [Work supported by a grant from NIH/NIDCD R01-DC006879].
The effects of off-frequency listening on the pitch of narrowband complexes.
Authors: Eckstein-Lilley Veronica , Berg Bruce
Published: October 2013

A pitch-matching paradigm was used to investigate the filtering properties of the auditory system. Listeners adjusted the frequency of a pure tone to match the pitch of a five-tone complex. Each complex had five equal amplitude sinusoidal components with equal frequency spacing (20 Hz). Five of six total conditions constituted a signal added in phase to any one of the five components, while the sixth condition added no signal. Two different signal levels were tested. Pitch matches were best described by the envelope weighted average of the instantaneous frequency, but only if the stimulus was first passed through the skirt of an auditory filter [V. Eckstein and B. G. Berg, J. Acoust. Soc. Am. 120, 3126 (2006)]. The results appear to be more consistent than those previously obtained with three-tone complexes. By using additional components and adding a signal at the edges of the signal complex, this study depicts a more complete inspection of the auditory filter shape, reduces pitch ambiguity, and eliminates nonlinearities. [Work supported by NSF.]
Statistical inference in the perceptual learning of non-native speech category.
Authors: Zhao Yuan
Published: October 2013

The study investigated the influence of category bias and category variance in English speakers’ learning of two lexical tone categories (high versus low). In Experiment 1, all subjects completed an AX discrimination pretest of adjacent tone pairs on a ten-step pitch continuum. They were then trained to associate a “low tone” with the meaning “pink” and a “high tone” with the meaning “blue” via a picture-sound association task. Subjects received either a biased training with a 5:1 ratio between high and low tones or an unbiased training with a 1:1 ratio between the two categories. It was then followed by an identification post-test. The results showed that poor pitch discriminators categorized significantly more sounds into the biased category. Good discriminators categorized significantly more sounds into the other category. Therefore, good discriminators have a novelty preference, while poor discriminators have a familiarity preference. In Experiment 2, subjects received either a small or a big variance training. The results showed that poor discriminators performed significantly better if trained with a big-variance input. However, good discriminators’ categorization performance was significantly better if trained with an input of a small variance. The study suggests that statistical cues are utilized differently among subjects of different auditory sensitivities.
A functional model of the slope sensitivity of the primary auditory cortex neurons in awake cats.
Authors: Ozawa Kenji , Koike Yoshikazu , Wakagi Hiromi , Sato Yu , Chimoto Sohei
Published: October 2013

Our measurement of single-spike responses to tone-burst stimuli showed that there are heterogenic response types of the primary auditory cortex (A1) neurons in awake cats: phasic neurons sensitive to stimulus amplitude-rise-slope at the onset of stimuli, sustained cells sensitive to the stimulus slope, and the steady state level at later time-phase of stimuli. In this study, in order to explain the behavior of the slope-sensitive phasic cells, an auditory model has been developed. The model consists of six stages: the inner hair cell (IHC), the primary auditory nerve (AN), the cochlear nucleus (CN), the inferior colliculus (IC), the medial geniculate body (MGB), and the A1 neuron. The model was implemented as a MATLAB program on a personal computer using Meddis’s model for IHC [R. Meddis, J. Acoust. Soc. Am. 79, 702-711 (1986)] and Maki’s model for AN, CN, and IC [Maki et al., J. Acoust. Soc. Jpn. 60, 304-313 (2000)], while the models of MGB and A1 neuron were newly developed. The model output successfully replicated the physiological data in terms of response rise-time, maximum driven rate, and onset-latency corresponding to the slopes of input stimuli. [Work supported by KAKENHI 20300076.]
Improvement of scale factor for time-domain audio watermarking based on low-frequency amplitude modification.
Authors: Murata Harumi , Ogihara Akio , Iwata Motoi , Shiozaki Akira
Published: October 2013

The objective of this work is to improve the sound quality for the audio watermark method based on amplitude modification. We improve the sound quality by modifying the scaling curve smoothly. The audio watermarking method based on amplitude modification has been proposed by Lie as a prevention technique against copyright infringement. The watermark information is embedded into audio signals in the time domain. One-bit watermark information is embedded by modifying the differences of average-of-absolute-amplitude from three sections in a group of samples. At the section boundary points, the scaling factors are forced to 1.0 to ensure continuity. The factors are then progressively increased or decreased to the stable value near the center of section. However, when the adjacent sections were both increased or both decreased, signal discontinuities happen in this boundary point and cause the “click” sounds that are perceivable to human ears. In this paper, we aim to make the audio waveform continuous and smooth with the scaling factors of the section boundary points that are not set to be 1.0 but are determined according to the stable values of the adjacent sections. By the proposed method, sound quality can be improved in objective and subjective evaluations.
Applying acoustic ray tracing from the full wave equation solution to determine the nonair conduction pathways into the human head.
Authors: McNew Jared , Bellina Alessandro , O'Brien William D.
Published: October 2013

Flight deck crews are subject to high intensity sounds for prolonged periods of time. Even while using current hearing protection devices such as ear plugs, ear muffs, and helmets, hearing loss is still possible. It is likely that this hearing loss is due to the cochlear stimulation through pathways other than the usual air conduction pathways. In order to develop new hearing protection devices to prevent hearing loss due to these nonair conduction pathways, ray tracing from the full wave equation solution is being used to determine and visualize these dominant pathways. Numerical techniques for performing ray tracing are presented along with the validation of the method by comparing with Snell's law. For small changes in sound speed, angle of transmission errors are less than 20%. As the frequency of the incident pulse is increased, these errors are reduced as would be expected. Results of ray tracing performed on simulation data from two concentric fluid spheres as an approximation to the human head are also presented. Using this model, an insertion loss of 26 dB is observed. [Work supported by AFOSR FA9550-06-0128.]
Subjective ratings of sentences in clear versus conversational speech.
Authors: Ferguson Sarah Hargus , Kerr Emily E.
Published: October 2013

Investigations of the relationship between speech acoustics and intelligibility face a significant methodological challenge, especially when using materials produced by multiple talkers and/or in multiple speaking styles. Ideally, acoustic analyses comparing talkers or styles will use identical materials for each. In contrast, intelligibility studies using meaningful stimuli must use either different materials or different listener groups to avoid learning effects. The latter solution becomes prohibitive when the number of talkers is large, such as in the Ferguson clear speech database (2004). While talkers recorded different materials in each speaking style, materials were the same for all talkers. For word intelligibility, familiarization prior to testing can prevent learning effects. For sentences, however, different listeners would be needed for each talker, resulting in 41 listener groups. An alternative solution is to use subjective rather than objective measures. This project explored the feasibility of this approach. Young listeners with normal hearing rated the clarity of sentences produced by eight talkers from the Ferguson database. Rated clarity was significantly higher for clear versus conversational sentences and varied significantly among talkers. Further analyses will compare clarity to perceptual and acoustic measures such as vowel intelligibility, speaking rate, and vowel space characteristics.
Effects of talker and token variability on perceptual learning of dialect categories.
Authors: Pate John K. , Clopper Cynthia G.
Published: October 2013

Dialect classification is difficult for naive listeners, but perceptual learning tasks using sentence-length utterances have been shown to produce modest improvements in performance. The goal of the current study was to explore perceptual learning by naive listeners in a speeded dialect classification task with shorter word-length utterances. In a series of experiments, participants were trained in a two-alternative forced-choice dialect classification task (Cleveland versus Cincinnati) with feedback and were then tested in the same task with novel talkers and novel words without feedback to assess learning. Variability in the stimulus materials in the training phase, including the number of talkers from each dialect and the number of different tokens produced by each talker, was manipulated across experiments to determine how variation in the input affected perceptual learning of dialect categories. The results revealed that training materials consisting of utterances produced by multiple different talkers from each dialect with multiple different tokens produced by each talker led to a significant improvement in dialect classification performance compared to a baseline condition without training. These findings suggest that dialect classification performance can improve with training on short word-length utterances, but that robust dialect category learning requires high variability stimulus materials.
Channel segregation improves perception of speech with temporally desynchronized bands.
Authors: Kiefte Michael , Stilp Christian E. , Kluender Keith R.
Published: October 2013

Stilp et al. [J. Acoust. Soc. Am. 122, 2971 (2007)] investigated the intelligibility of temporally desynchronized bands of speech and concluded that listeners’s resilience to temporal asynchrony may possibly be due to differences in fundamental frequency (f0) across syllables which may help in segregating bands. This hypothesis is tested in two experiments in temporally desynchronized bands but with spectral manipulations designed to either aid or inhibit stream segregation. Seven-syllable sentences were synthesized at three different speaking rates and processed by four nonoverlapping 1/3-octave filters. Onsets of the lowest- and highest-frequency bands were parametrically delayed. In Experiment 1, f0 in delayed bands was uniformly elevated using pitch-synchronous overlap add synthesis. In the control condition (no f0 manipulation), intelligibility was nonmonotonic with delay across speaking rates with local minima corresponding to the duration of one syllable. The two-speaker manipulation decreased spectral similarity across bands making intelligibility more resilient to temporal distortion. In Experiment 2, f0 contours were flattened. Intelligibility was uniformly poorer than in the control condition; the increased spectral similarity compromised listeners’ ability to segregate information from band pairs at various delays. Acoustic measures of potential information, absent explicit linguistic information, reinforce the strong relationship between spectral predictability and intelligibility. Overall, results provide support that channel segregation plays an important role in the perception of temporally desynchronized bands. [Supported by NIDCD]
Evaluation of two voice stress analyzers.
Authors: Hollien Harry , Harnsberger James
Published: October 2013

The purpose of this study was to evaluate two commonly used voice stress analyzers: NITV’s computer voice stress analyzer (CVSA) and Nemesysco’s layered voice analysis (LVA) system. In both cases, a speech database was used, which contained materials recorded (1) in the laboratory, while highly controlled deceptive and shock induced stress levels were systematically varied and (2) during a field procedure. Subjects were 24 males and females (age range 18–63 years) drawn from a population representative of the United States. All held strong views on an issue and were required to make sharply derogatory statements about it. The systems were then evaluated in a double blind study using two sets of examiners: (1) two UF scientists trained/certified by the manufacturers and (2) either three experienced CVSA operators or two LVA instructors provided by the manufacturer(s). The results for both devices showed that the “true positive” (or hit) rates ranged from chance to somewhat higher levels—50% to 65%—for all conditions/types of materials (stressed-unstressed, truth, or deception). However, the false positive rate was just as high—often higher. Sensitivity statistics demonstrated that these systems operated at about chance levels. [Work supported by Counterintelligence Field Agency, DoD.]
Effects of acoustic transformation on cross-modal speech information and audiovisual gain II.
Authors: Dias James W. , Lachs Lorin
Published: October 2013

Auditory and visual perceptual processes interact during the identification of speech sounds. Some evaluations of this interaction have utilized a comparison of performance on audio and audiovisual word recognition tasks. A measure derived from these data, R, can be used as an index of the perceptual gain due to multisensory stimulation relative to unimodal stimulation. Recent evidence has indicated that cross-modal relationships between the acoustic and optical forms of speech stimuli exist. Furthermore, this cross-modal information may be used by the perceptual mechanisms responsible for integrating disparate sensory signals. However, little is known about the ways in which acoustic and optic signals carry cross-modal information. The present experiment manipulated the acoustic form of speech in systematic ways that selectively disrupted candidate sources of cross-modal information in the acoustic signal. Participants were then asked to perform a simple word recognition task with the transformed words in either auditory-alone or audiovisual presentation conditions. It was predicted that audiovisual gain would be relatively high for those transformations in which the relative spacing of formants was preserved but would be nonexistent for those transformations that destroy the relative spacing of formants. The results are discussed in terms of existing theories of audiovisual speech perception.
Effects of lexical status within phonetic categories.
Authors: Soleimany Sam , Sawusch James
Published: October 2013

Previous work has shown an interaction between lexical knowledge and phonetic categorization. For example, Ganong (1980) showed that ambiguous initial phonemes were identified so that the resulting syllable was a word. While previous studies have emphasized lexical influences at the phonetic category boundary and their time-course, the present work examines lexical influence effects within categories. Changes in within category discrimination for word and nonword tokens were examined with an ABX discrimination paradigms. Signal detection analysis was employed to determine whether changes in rating responses within a phonetic category were attributable to changes in sensitivity, bias, or both. While changes in identification and discrimination between phonetic categories may be attributable to postperceptual biases, changes within categories may not be. Results will be discussed in regard to their impact on theories of word recognition and whether lexical influences are prelexical, postlexical, or both.
Speech perception in a language-trained chimpanzee (Pan troglodytes).
Authors: Heimbauer Lisa , Beran Michael , Owren Michael
Published: October 2013

After decades of research, the question of whether humans perceive spoken language using a specialized “speech mode” remains unresolved. Studies in nonhumans suggest that animals perceive phonemic contrasts much as humans do, but involve subjects trained for thousands of trials on single discriminations. This work reports initial speech perception results from Panzee, a chimpanzee (Pan troglodytes) reared by humans speaking to her as they would to a child and also training her to use graphical wordlike “lexigrams.” Panzee comprehends approximately 126 spoken words, documented through a procedure in which a digitally presented spoken word is matched to one of four lexigrams presented on a monitor. First experiments have compared performance with natural digitized versions of 24 spoken words to synthetic LPC-based replicas and to whispered versions. Using a different subset of eight test words within each of three 96-trial sessions showed comparable mean performance for natural (83.3%), synthesized (82.5%), and whispered (78.5%) versions. Percent-correct performance on 24 trials representing the first time a given test word was heard was also comparable for synthesized (79.2%) and whispered (78.5%) sounds. The possibility that Panzee is showing speech-mode perception will be tested in experiments with noise-vocoded and sine-wave speech. [Work supported by NICHD.]
Descriptors of aural connectivity: Architectural enclosure features and acoustical qualities.
Authors: Okcu Selen , Zimring Craig , Ryherd Erica
Published: October 2013

In intensive care unit (ICU) settings, auditory monitoring of alarm and nonalarm sounds is one of the key nursing functions. In the ICU sound environments there are many different noise sources, some producing more than one tone to communicate different conditions. In a sound environment with many different sound layers offering different sound qualities and sound levels nurses are expected to distinguish the sounds from each other and identify critical ones. Earlier studies showed that effective auditory monitoring requires recognition, localization, and immediate reaction to critical sounds to improve patient safety, nurse work efficiency, and reduce nurse burnout. This study explores the nurse task specific auditory monitoring experience by documenting the nurse auditory tasks and systematically mapping related sounds. This is analyzed as aural connectivity, a network measure that reflects the overall pattern of where users can recognize and localize the critical sounds in architectural enclosures. It is hypothesized that enclosure types shape sound environments that affect user auditory experience. This paper defines the metrics for enclosures and associated acoustical qualities by looking at two ICU settings and discusses the preliminary results regarding the relationship between qualities of sound environment and architectural enclosures.
Defining perceptual requirements of real-time acoustic auralizations.
Authors: Gedemer Linda , Braasch Jonas
Published: October 2013

Acoustic modeling evolves from static sources and receivers to dynamic sources and receivers as computational speeds allow for faster renderings. Current research has focused on creating highly accurate dynamic models with the hope of producing accurate auralizations but still struggles with the trade-off between accuracy and available processing speed. The question remains as to how accurate do these models need to be if the receiver (listener) moves dynamically through the acoustic model. The focus of this paper is to address the accuracy required of dynamic models and/or auralizations from a perceptual standpoint. From this, the necessity for creating models that are perhaps more accurate than required will be considered.
Auditory-visual equivalence learning by marine mammals.
Authors: Schusterman Ronald J. , Lindemann Kristy , Reichmuth Colleen
Published: October 2013

Forty years ago Sam Ridgway began pioneering research on the brain and sensory systems of marine mammals. Our paper honoring Sam reviews studies on the abilities of dolphins and California sea lions to integrate and classify information coming from different sensory channels in cross-modal matching-to-sample tasks. Following echoic recognition of an object, a dolphin that is permitted to investigate objects visually but not echoically can readily learn and remember that earlier reflected sound cues are related to the currently reflected light cues. A California sea lion has shown that it can also integrate auditory-visual information into something resembling equivalence classes. In these studies the sea lion was taught to relate an array of individual auditory signals (e.g., “ringing bell” and “wailing siren”) to several individual members of two previously established visual equivalence classes (“numbers” and “letters”). Following the training phase it was later determined that the sea lion had knowledge of untrained transitive relationships between the auditory signals and the remaining members of the two visual equivalence classes. Such findings suggest that these marine mammals have neural pathways connecting auditory and visual associative brain areas in the construction of long lasting integrated or global representations.
Sound attenuation coefficients in sandy and silty seabottoms from low- to mid-frequency field measurements in shallow water.
Authors: Zhou Ji-Xun , Zhang Xue-Zhen , Chotiros Nicholas P.
Published: October 2013

The debate on the sound speed dispersion and the frequency dependence of sound attenuation in sediments has persisted for decades, mainly due to the lack of sufficient experimental data in the low- to mid-frequency speed/attenuation transition band and the limitations of existing models. This paper analyzes and summarizes the LF measurements, conducted at 20 locations in different coastal zones around the world, that have resulted in the identification of nonlinear frequency dependence of sound attenuation in sand-silt seabottoms for a frequency range of 50–2000 Hz [Zhou and Zhang, J. Acoust. Soc. Am. 117, 2494 (2005); 119, 3447 (2006)]. The resultant LF sound speed and attenuation can be described equally well by the Biot-Stoll model, the Chotiros BICSQS model, and the Buckingham VGS model. However, a combination of the LF field-inverted sound speeds and attenuations with data from the SAX99 and SAX04 measurements over a broadband of 50–400 000 Hz cannot simultaneously be matched by any of these models with one set of adjustable input parameters; these models either underestimate broadband dispersion or overestimate LF attenuation. Comments on the LF field-derived data, such as “speed-attenuation coupling,” are made. The possible reasons for the broadband data-model mismatch are discussed. [Work supported by the ONR.]
The use of coded signals in the measurement of diffusing surfaces.
Authors: D’Antonio Peter , Cox Trevor
Published: October 2013

Following the introduction of quantifiable number theoretic, reflection phase grating diffusing surfaces by Schroeder [J. Acoust. Soc. Am. 57, 149–151 (1975)], several approaches have been used to measure and characterize them. In the early 1980s, time delay spectrometry, developed by Heyser, was used to measure the temporal, spatial, and spectral responses of diffusing surfaces. Once maximum length sequences were introduced as an excitation signal, they were adopted in the late 1980s. These explorations led to the creation of a measurement standard for the diffusion coefficient, which is the circular autocorrelation of the third octave polar responses. The limitations of periodic number theoretic surfaces, namely, bandwidth, grating lobes, and flat plate effects, were addressed by computer optimization, which combined the power of boundary element and multi-dimensional minimization techniques, and modulation. Both one-dimensional (phase variation in one direction) and two-dimensional (phase variation in two orthogonal directions) surfaces were characterized. The data collection and reduction procedures will be reviewed with several examples of diffusing surfaces. A summary of this research can be found in the work of Cox and D’Antonio [Acoustic Absorbers and Diffusers: Theory, Design and Application (Spon, 2004)].
Orthogonal coding sequences for multiple-source study of outdoor noise propagation.
Authors: Waddington David , Angus Jamie
Published: October 2013

This paper describes recent developments in the design of a coded acoustic signal for the study of outdoor sound propagation. Low signal-to-noise ratio is a common limitation in outdoor propagation investigations, and time variance of the air limits the amount of averaging. Recent work by the authors presented a coded acoustic signal approach consisting of a frequency carrier biphase-modulated by a specially designed pseudorandom code sequence to overcome these limitations. The “inner and outer” code sequence was specially designed for environmental sound propagation investigations, combining simultaneous fine time resolution and large range ambiguity, together with an ability to average and probe the propagation path. This approach is extended to multiple acoustic sources for the study of outdoor sound propagation, with the design of signals using orthogonal correlation sequences. The results illustrate that combinations of orthogonal coding and multiple carrier frequencies can permit continuous sound speed measurements in numerous vertical and horizontal directions and might therefore be useful for acoustic tomographic applications.
Multimodal coded excitation in medical ultrasonic imaging.
Authors: Wan Yayun , Ebbini Emad
Published: October 2013

Coded excitation has been used successfully in medical ultrasound to improve the signal to noise ratio and penetration depth. Furthermore, inverse filtering techniques have been shown to improve axial resolution without significant loss in contrast due to the range sidelobe phenomenon. With the next generation ultrasound scanners, the opportunity exist for transmitting multiple codes simultaneously, with or without focusing, into the region of interest. Using appropriate 2D inverse filtering on receive, it is possible to control the axial and lateral resolutions of the imaging system while maintaining an acceptable tradeoff with image contrast. In this paper, we review recent results on 1D and 2D inverse filtering of beamformed echo data resulting from insonification of speckle-generating cyst phantoms with multiple codes. We demonstrate that (regularized) inverse filtering can be derived from the space-time array patterns. Furthermore, we demonstrate the feasibility of reconstructing multiple image lines in parallel to increase the frame rate in 2D and 3D imaging.
Pseudorandom sequences for binary amplitude diffusers.
Authors: Dadiotis Konstantinos , Angus Jamie A. S. , Cox Trevor J.
Published: October 2013

Binary amplitude diffusers create a mixture of absorption and diffusion and are used to improve room acoustic conditions. They require a two dimensional binary pseudorandom array to state where absorbing and reflecting patches should be. Consider the binary array to have dimensions A×B. Ideally, A and B should be similar to maximize performance. Such an array can be produced by folding a one dimensional sequence into a two dimensional array using the Chinese Reminder Theorem. This theorem requires A and B to be coprimes. However, there is a limited set of optimal one dimensional sequences that can be generated from number theory with suitable length AB. In the past maximum length sequences have been used because they display desirable autocorrelation characteristics, but their period is 2m−1, where m is an integer and so there is a limited number available. Consequently, it is necessary to examine other ways of generating optimal two dimensional arrays. This paper looks into other number theoretic sequences, and other construction techniques for forming binary arrays. The suitability of these for constructing two dimensional binary amplitude diffusers is examined. Boundary element modeling is used to evaluate their performance.
Four buildings converted to performance spaces.
Authors: Woolworth David
Published: October 2013

This paper documents the conversion of two metal buildings and two brick buildings into performance spaces: a metal seafood restaurant into a church sanctuary, a cotton gin into an amplified music venue, a 1928 power house into a black box and gallery, and an office complex and restaurant into a 1200-person capacity amplified venue and movie theater. The challenges and trade offs will be discussed, and the coordination of room acoustics, mechanical noise, and sound systems. Methods of control of amplified low end are addressed for each venue in terms of isolation and absorption.
Retrofitting a performance space: Everyman Theatre.
Authors: Fischer Julie
Published: October 2013

The new home of the Everyman Theatre company opened in 1910 as a Vaudeville house and has had many reincarnations throughout its lifetime, including a parking garage and 1550-seat movie theater. In its latest reincarnation, the space under renovation was a large movie theater. A challenge for the project team has been to take this configuration and turn it into something that fits the needs of the Everyman Theater company. The current renovation is underway and will include a large studio theater, with approximately 250 seats and several different configurations including a typical end stage, thrust stage, and arena stage, a Blackbox theater, and support spaces such as classrooms, offices, and rehearsal space. One of the main issues related to the pre-existing space is its current structure. The current design has two theaters stacked vertically and there is concern that noise and vibration from activities from the upper theater could disturb occupants in the lower theater. Upgrading the structure is complicated by the location of the building which sits directly above a metro tunnel and therefore has strict weight requirements. The approach currently proposed is to include a floating concrete floor and sound barrier ceiling. [Chris Pollock (SMW), Rima Namek and Diane Cho (Cho Benn Holback), Millie Dixon and John Coyne (Theatre Projects), and Vincent Lancisi (Everyman Theatre).]
Multithread implementation for calculating room impulse responses.
Authors: Ahnert Wolfgang , Feistel Stefan , Miron Alexandru
Published: October 2013

Modern simulation programs are committed to calculate full impulse responses instead of doing simple mapping presentations. The main problem here is how to obtain such IRs quickly but accurately. The usual algorithm simulates sound propagation by tracing a statistical ensemble of acoustic rays. In this contribution, we deal with a new ray-triangle intersection algorithm, based on a three-dimensional grid scheme, that provides significant performance improvement for ray tracing. We report about the introduction of a uniform grid ray-tracing algorithm, which is optimized for acoustic models and can be up to five times faster (standard Pentium 4) than hierarchical space decomposition. Furthermore, the algorithm has been parallelized, taking advantage of the inherently independent propagation of distinct rays. On multicore processors, the computation employs a variable number of thread tracing groups of rays in parallel. The speed increase is almost linear with the number of cores. An overview is given to explain this approach. To support this fast calculation, an additional algorithm derives a suggested particle number and length from the mean free path properties of the room and the desired detection rate at the receiver. So time-consuming guesses about the selecting particle numbers will be avoided.
Swept sine against maximum length sequences in room acoustics with music signals as background noise.
Authors: Paulo Joel , Bento Coelho José
Published: October 2013

The Swept Sine and the MLS techniques are very popular in room acoustic measurement set-ups. Advantages and disadvantages of both methods have been well investigated and can be found in the literature. However, information regarding the performance of these techniques in the presence of high background music levels is scarce. Since the estimation of the room impulse response is based on the correlation between signals, the likelihood between the test signal and the music contents plays an important role on the results’ accuracy. This paper explores these issues by taking into account the semantic information of the music signals when used as disturbance. The method used for the assessment of the gain between the two techniques consists of splitting each frame into segments and applying a weighting function depending on a likelihood function. The features used for the likelihood function are the rms value of each segment, spectral energy envelope relation, bandwidth and harmonic structure. Several examples are presented for comparison of the performance of the Swept Sine and the MLS techniques. Advantages and disadvantages of each technique are discussed for music signals as noise.
Time reversal acoustic receiver.
Authors: Fillinger Laurent , Sinelnikov Yegor , Sutin Alexander , Sarvazyan Armen
Published: October 2013

The time reversal acoustics (TRA) principle has been employed for focusing ultrasonic waves in various biomedical and industrial applications. TRA-based ultrasonic transmitters have numerous advantages over conventional methods of ultrasound focusing such as an ability to effectively focus and steer focal spot with a transmitter comprising even a single piezoelectric transducer. In this study it is shown that such TRA focusing systems with a small number of transducers can also work in the receiving mode and detect both active and passive targets with a spatial resolution close to a half wavelength. The feasibility tests were conducted using two types of TRA receivers. The first TRA receiver comprising a water-filled reverberator with a few 600 kHz piezoceramic elements attached was used for the detection of the signal radiated by a miniature (1 mm) emitter in a water tank. It was shown that the TRA receiver can detect the position of this point emitter with accuracy better than 1 mm. The second test was conducted with an acoustic reverberator made of series of thin randomly shaped membranes and a single disk transducer with a frequency of 3.3 MHz. This TRA receiver allowed the localization of a reflecting target illuminated by a plane acoustic wave with resolution also better than 1 mm.
Ultra-wide-band filter for noise control.
Authors: Kushwaha Manvir
Published: October 2013

Extensive band structures are performed for two-dimensional periodic arrays of rigid stainless steel cylinders in air, with Bloch vector being perpendicular to the cylinders. We embark on the opening up of a complete acoustic band gap—independent of the polarization of the wave and of the direction of propagation. In addition, we propose the fabrication of a multiperiodic system in tandem that could create a huge hole in sound within the human audible range of frequencies.
An automated calibration system for telematic music applications.
Authors: Braasch Jonas
Published: October 2013

The complex polyphonic nature of music typically has stricter requirements for sound-pressure level calibration than back-and-forth natured speech dialogs. An onsite sound engineer typically controls the levels for electroacoustically enhanced music performances. In telematic applications, however, this task can easily become difficult because the sound engineer only has restricted access to the remote sites. This makes it difficult to balance the overall sound. For this reason, an automatic calibration system was developed. The system works with closely captured microphone signals, which are needed to avoid echo-feedback between two or more remote telepresence sites. For calibration purposes, a small-aperture pyramidal five-microphone array is mounted in the center of each remote site. The microphone array tracks the positions and sound-pressure levels of the individual instruments, based on an algorithm that can extract this information in the presence of concurrent sound sources [Braasch and Tranby, 19th ICA, Madrid, Spain (2007)]. The extracted data are then transmitted via OpenSound Control to the remote site(s) for accurate spatial reproduction of the original sound-source positions. The signals are calibrated such that they evoke the same sound-pressure levels at the microphone array of each site.
Computation of three-dimensional, pulsed, nonlinear acoustic wavefields from medical phased array transducers in very large domains.
Authors: Huijssen Jacob , Verweij Martin D. , De Jong Nico
Published: October 2013

For the optimization and development of medical ultrasound transducers and imaging modalities, the iterative nonlinear contrast source (INCS) method has been developed. This numerical method predicts the nonlinear acoustic pressure field generated by a pulsed plane source with an arbitrary aperture and propagating in a three-dimensional tissuelike medium that extends over a very large domain of interest. The INCS method obtains the acoustic pressure from the nonlinear acoustic wave equation by treating the nonlinear term as a contrast source. The full nonlinear wave field is then found by iteratively solving the linearized wave problem using a Green's function method. By employing the filtered convolution method discussed in a companion paper, accurate field predictions are obtained at a discretization approaching two points per wavelength or period of the highest frequency of interest. In this paper, very large-scale nonlinear field profiles are presented for transducers with cylindrical as well as phased array geometries excited with a pulsed waveform having a center frequency of 1–2 MHz. Comparison with the results obtained from models of reduced complexity shows that in all cases the INCS method accurately predicts the nonlinear field. [Work supported by the STW and the NCF.]
Impact of segmentation rules on the rhythm metrics.
Authors: Stojanovic Diana
Published: October 2013

Rhythm of speech in the literature of the past decade is quantified using measures of durations: first order, such as mean duration and percentage of the total sample, or second order, such as variance and pairwise variability index. Results of different studies, however, vary even when the same materials and speech mode are used and imply uncertainty of cross-study comparisons. Allen (1978) discussed the issue of validity (the criteria used for segmenting) and reliability (how accurately these criteria can be applied) on duration studies. The current paper examines how different segmentation criteria affect the five most commonly used rhythm metrics. All metrics are computed on the segmented sample of the read story (The North Wind and the Sun) for three speakers each of American English, Indonesian, and Serbo-Croatian. Different segmentation criteria were applied to (1) borders between voiceless plosives and vowels and (2) borders between vowels and trills. As a result, durations of some segments were changed in the range of 10–50 ms. The results show how different metrics are affected by the choice of two segmentation rules at two different speech rates. Implications for cross-linguistic studies are discussed.
Acoustic correlates of vocal effort.
Authors: Tasko Stephen , Parker Madalyn , Hillenbrand James
Published: October 2013

This study evaluated the influence of open quotient (OQ), fundamental frequency (F0), and intensity on listener ratings of vocal effort for isolated synthetic vowels. Glottal waveforms (duration: 1000 ms) were synthesized using a model developed by Rosenberg [J. Acoust. Soc. Am. 49, 583–590 (1971)]. Signals were generated using six different OQs ranging from 0.2 to 0.7 at each of eight distinct F0 values. Four F0 values were in a typically male range (100–142 Hz) and four in a typically female range (178–253 Hz). The glottal waveforms were passed through formant resonators with formant frequency settings characteristic of the vowel /a/. Signals were randomly presented at 80, 83, and 86 dBA to 20 listeners who used visual analog scales to rate each sample on dimensions of vocal effort, loudness, pitch, breathiness, and naturalness. Mean vocal effort ratings showed a weak negative association with OQ (r=−0.34) and weak-moderate positive associations with F0 (r=0.45) and intensity (r=0.55). Multiple regression analysis revealed that a linear combination of the three variables accounted for 68% of the variance in vocal effort ratings. These results suggest that the perception of vocal effort relies on multiple acoustic cues.
Perceived prosody: Phonetic bases of prominence and boundaries.
Authors: Cole Jennifer , Goldstein Louis , Katsika Argyro , Mo Yoonsook , Nava Emily , Tiede Mark
Published: October 2013

In comprehending speech, listeners are sensitive to the prosodic features that signal the phrasing and the discourse salience of words (prominence). Findings from two experiments on prosody perception show that acoustic and articulatory kinematic properties of speech correlate with native listeners’ perception of phrasing and prominence. Subjects in this study were 114 university-age adults (74 UIUC + 40 Haskins), monolingual speakers of American English who were untrained in prosody transcription. Subjects listened to short recorded excerpts (about 20 s) from two corpora of spontaneous and read speech (Buckeye Corpus and Wisconsin Microbeam Database) and marked prominent words and the location of phrase boundaries on a transcript. Intertranscriber agreement rates across subsets of 17–40 subjects are significantly above chance based on Fleiss’ statistic, indicating that listeners’ perception of prosody is reliable, with higher agreement rates for boundary perception than for prominence. Prosody perception varies across listeners (both corpora) and across speakers (WMD, where perceived prosody varies for the same utterance produced by different speakers). Acoustic measures from stressed vowels (Buckeye: duration, intensity, F1, F2) and articulatory kinematic measures (WMD) are correlated with the perceived prosodic features of the word. [Work supported by NSF.]
The development of tonal duration in Mandarin-speaking children.
Authors: Yang Jie , Diehl Randy , Davis Barbara
Published: October 2013

Previous research found that the duration of segments decreases as children grow older. The development of suprasegmental duration, however, has not been explored. The present study investigated developmental changes in duration of the four Mandarin tones. 5-, 8-, and 12-year-old monolingual Mandarin-speaking children and young adults participated in the study. Tone durations were measured in participants’ production of monosyllabic target words elicited by picture identification tasks. The results were as follows (1) For each tone category, tone duration and variability decreased with age: 5- and 8-year-old children showed significantly longer durations than adults. Tone durations in 12-year-old children approximated adult values. (2) Despite longer durations, adultlike duration patterns across tone categories existed in all children: dipping tones were the longest, followed by rising and level tones, with falling tones being the shortest. (3) Duration differences between the rising and dipping tones became larger as children grew older. The results may be indicative of the general maturation of laryngeal control over age. Although 5- and 8-year-old children have already established lexical contrasts of tone, adultlike phonetic norms are still in the process of development. The developmental data also provide support for a hybrid account of speech production from a suprasegmental perspective.
Mitigation of speckle noise due to laser Doppler vibrometer motion across a vibrating target.
Authors: Burgett Richard , Aranchuk Vyacheslav , Sabatier James
Published: October 2013

Laser Doppler vibrometers (LDVs) are widely used for remote vibration measurements. In many applications, such as measurements with an LDV on a moving platform or measurements with a continuously scanning beam, an LDV beam moves across a target. The motion of a laser beam across the target generates noise at the LDV output due to dynamic speckles—speckle noise. Speckle noise is caused by phase fluctuations of laser speckles and by Doppler signal dropouts and increases with the speed of the beam. It has been observed that the noise floor increase due to speckles follows a logarithmic curve as the speed increases. Of course as the speed increases, the dwell time over the target area decreases, which limits the signal of interest. This smaller sampling of the vibration signal, combined with the increase in noise due to speckles, places a severe limit on the scanning speed. A way is being investigated to take advantage of this nonlinear increase in noise with respect to speed by using time multiplexing of multiple LDV beams to increase the length of time the LDV samples the vibrations of the target area, thereby allowing the target signal to be detected above the noise floor.