bluuebunny/crossref_metadata_2025 huggingface dataset:
Composite wavelet transform and frames
Authors: Trappe Wade , Lakey Joseph D.
Published: October 2005

Motivated by the human auditory system, a new signal transform is presented which models the way humans hear. Cochlear processing acts like a constant bandwidth bank of filters in the low frequency range but is of proportional bandwidth at higher frequencies. This new transform, which we call the composite wavelet transform, is better able to model this process than standard signal processing techniques such as the short-time Fourier transform (STFT) and the continuous wavelet transform (CWT). The composite wavelet transform in fact provides a signal analysis tool that is able to examine signals with competing signal structures whereas the STFT and the CWT do not. In order to insure stable signal recovery, the theory of frames is examined. An overview of the theory of frames will be given and wavelet and Gabor frames will be discussed. Recent work on frames for the composite wavelet transform will also be presented. Finally, some applications of frames, especially to speech signal processing will be considered.
Shrinking illusion of speech duration caused by noise/gap replacement
Authors: Tsuzaki Minoru , Kato Hiroaki
Published: October 2005

To investigate the process of perceptual resynthesis of an ‘‘auditory object,’’ the apparent duration of a noise-replaced speech signal was measured. Subjects adjusted the duration of synthesized speech to subjectively match that of the target speech. When a portion of the target speech was replaced by a noise burst, the duration was perceived to be shorter than when no replacement was made. Besides this effect, the apparent duration of speech segments was found to be affected by their deviation from the internal template. The greater the physical duration of the target deviated from the ‘‘naturally’’ spoken token, the greater the degree of underestimation became. Interestingly, these two effects were independent of each other. This independence was replicated in the second experiment which tested wider range of durational deviations. Again, the apparent duration of noise-replaced speech was shorter than when no replacement was made. The apparent duration of gap-replaced speech was also measured in this experiment. The degree of underestimation was far greater for the latter case. These findings suggest that a shrinkage of duration occurs in the process of resynthesizing an ‘‘auditory object’’ where the process is assumed to be at a different stage than that utilizing speech-specific schema.
Mechanisms underlying nonmonotonic Weber functions in forward-masked intensity discrimination
Authors: Clement Bart R. , Schlauch Robert S.
Published: October 2005

Nonmonotonic growth in threshold for intensity discrimination has been noted for pure tones preceded by intense sinusoidal maskers; thresholds are highest for 40–60 dB SPL standards, producing what has often been described as a ‘‘midlevel hump.’’ Both physiologic and cognitive factors have ben suggested as mechanisms underlying this phenomenon. Three subjects received four conditions designed to examine this issue. A 2IFC paradigm assessed intensity discrimination of 1000-Hz, 100-ms sinusoids. A no-masker condition provided a baseline. Three forward-masked conditions were implemented: An ipsilateral condition that typically produces a mid-level hump; a contralateral condition in which the hump is typically absent or greatly reduced; and a binaural condition presenting these two maskers simultaneously (contralateral stimulation was 13 dB greater than ipsilateral stimulation; listeners perceived only the contralateral masker). It was argued that if this condition produced a hump despite a perceptually absent ipsilateral masker, a sensory explanation would be suggested. Conversely, data mirroring the no-masker or contralateral results would be consistent with a cognitive hypothesis. Results suggest a strong sensory component, with only a small portion of the effect having a possible cognitive origin. Data will be compared to those of related literature. [Work supported by Bryng Bringelson Foundation.]
Short-term temporal integration at high-frequencies: Evidence for the influence of peripheral nonlinearity?
Authors: Oxenham Andrew J. , Moore Brian C. J.
Published: October 2005

Temporal integration in the auditory system can be modeled by a compressive nonlinearity followed by a temporal integrator. For a fixed integrator, the more compressive the nonlinearity, the greater the effective integration. Compressive nonlinearity on the basilar membrane is marked at medium levels, but is reduced at low and high levels. To assess whether this influences temporal integration, thresholds of a 6.5-kHz sinusoid were measured in the presence of a long-duration broadband-noise masker as function of signal duration, using three different noise spectrum levels, −10, 20, and 50 dB (re: 20 μPa). Signal level dB was plotted against log(time) for each level. For signal durations between 20 and 200 ms, there was no significant difference in the slopes of the three functions. However, between 1 and 10 ms the slope of the function for the mid-level masker was steeper by a factor of between 1.5 and 2. This is consistent with the greater mid-level compression on the basilar membrane, if one assumes a fixed short-term temporal integrator. Performance in other psychoacoustic tasks, including differences between normal-hearing and hearing-impaired listeners, is also discussed in terms of peripheral processing. [Work supported by the MRC, EPSRC, and Meridian Audio (UK).]
Application of confidence intervals and joint confidence regions to the estimation of psychometric functions
Authors: Hawley Monica L. , Colburn H. Steven
Published: October 2005

A mathematical description of a psychometric function with two free parameters is fit to fixed-increment data by a nonlinear gradient search technique that incorporates a weighted least squares algorithm. The statistical confidence in the parameter estimates is considered by comparing results from standard confidence interval analysis with those from joint confidence region analysis. Confidence intervals estimate the variability of each parameter alone, ignoring the interaction between the parameters, whereas the joint confidence region gives the confidence in the joint estimation of both parameters together. In our study, both analyses were applied to fixed-increment data from a variety of tests (interaural time and intensity discrimination and N0Sπ binaural detection) and subjects for two-interval forced choice experiments. Results show that the parameter estimates are affected primarily by the data collected near the midpoint which is not surprising since the parameters in the model are the slope and translation at the midpoint. However, the joint confidence region analysis shows that the statistical confidence of the parameter estimates are greatly affected by data collected at levels that give performance near chance and near perfect. [Work supported by NIDCD (Grant DC00100).]
Temporal symmetry in auditory perception and the auditory system
Authors: Irino Toshio , Patterson Roy D.
Published: October 2005

When a repeating damped exponential is used to modulate a sinusoid or noise, the carrier component of the perception is suppressed. When these ‘‘damped’’ sounds are reversed in time, producing ‘‘ramped’’ sounds, the carrier is restored. [R. D. Patterson, 1419–1428 (1994); M. A. Akeroyd and R. D. Patterson, 2941(A) (1994)]. We report matching experiments designed to measure the perceptual asymmetry between damped and ramped sounds directly. Listeners compared the relative strengths of the transient and continuous components in pairs of damped and ramped sounds, and the damped half life was varied to produce a match in the relative strengths of the transient and continuous components of the two sounds. For sinusoidal and noise carriers, the half life of the damped sound has to be, respectively, 5 and 2.5 times that of the corresponding ramped sound to produce the match. A ‘‘delta-gamma’’ theory of intensity enhancement around acoustic transients [T. Irino and R. D. Patterson, 2943(A) (1994)] is used to explain the perceptual asymmetry. The theory is also used to compare the asymmetry produced by different auditory models.
Infinite-Impulse-Response filter models of the head-related transfer function
Authors: Kulkarni Abhijit , Colburn H. Steven
Published: October 2005

Directional-transfer-functions (DTFs) derived from head-related transfer functions (HRTFs) measured from human subjects in an anechoic environment were modeled using auto-regressive (AR) and auto-regressive moving-average (ARMA) modeling techniques. The HRTFs, which were provided by Dr. Fred Wightman (Wightman and Kistler, 1989), were decomposed into a directional-transfer-function (DTF) and a mean-function (computed as the mean of all measured HRTFs for that subject). Model reconstructions of the HRTFs was done under the assumption that HRTFs are minimum-phase functions. The AR estimator which is derived from the theory of linear prediction, corresponds to an all-pole model of the DTF. The ARMA model extends the all-pole model to a pole-zero model and is derived by posing a weighted-least-squares formulation of the modified-least-squares problem proposed by Kalman (1958). The validity of each model was assessed psychophysically using a 4I, 2AFC paradigm. Using a 80-ms white noise stimulus and testing a different position on each trial, for a 25-pole all-pole model and a 6-pole 6-zero pole-zero model, subjects performed close to chance for most of the 13 representative positions tested. [Work supported by NIDCD (Grant DC00100).]
Measurement of localization ability using virtual audio cues and visual target acquisition
Authors: D’Angelo William R. , McKinley Richard L. , Perrott David R. , Hettinger Larry
Published: October 2005

Visual target detection was used to demonstrate the ability of subjects to localize with a virtual audio display. The testing facility consisted of audio presented over headphones with and without localization cues and a projection screen bearing the visual scene. In each trial the visual target appeared as one pixel at a random location of the screen. The target grew in size until found by the subject. The sound cue came on at threshold level and became progressively louder until the target was found. The audio cue was pink noise while the visual targets consisted of aircraft icons. Subject reaction time was measured from target onset to target detection. As indicated by the preliminary data, the localized audio condition faithfully provided the location of the target resulting in reaction times significantly better than with the nonlocalized audio cue.
Aurally aided visual search under static and moving conditions
Authors: Ferguson Debra L. , Perrott David R.
Published: October 2005

Auditory localization of static and dynamic targets in the presence or absence of an (broadband noise) auditory cue was examined in a two-alternative, forced-choice paradigm. The location of the target varied relative to the subject’s initial fixation point. The right and left hemispheres were tested at 45, 90, and 135 deg azimuth on the zero degree elevation plane. Also tested were zero and 180 deg locations. The range of target velocities was 0, 10, 20, or 40 deg per second, with direction of travel and velocity randomized in each trial. Each main effect was significant as well as two-way interactions between condition×location and condition×velocity. These results suggest the utilization of spatially correlated auditory cues under static and moving conditions, particularly in the posterior field and at higher velocities. [Work supported by NSF.]
The relationship between tone frequency and perceived elevation under headphone listening conditions
Authors: Musicant Alan D.
Published: October 2005

The relationship between pure tone frequency and perceived inheadelevation was investigated. Stimuli were pure tones ranging from 1000 to 1200 Hz in 1000 Hz steps. Insert earphones were utilized for presentation. Subjects were instructed to indicate, on two separate diagrams, the image location of the stimulus after presentation of a brief tone. Dimensions were left–right, front–back and elevation. All stimuli were presented diotically, that is with equal intensity to the two ears. Tones were presented at approximately 70 phons with a variation of ±3 dB. Perceived in head location was found to vary systematically in the elevation dimension. Tones with lower frequencies were perceived as lower in elevation than tones of higher frequency up to about 9 kHz. Stimuli with frequencies of 10 kHz and higher were perceived as lowerinelevation relative to the perceived elevation of the 8- and 9-kHz stimuli. Results of this experiment were compared to results from free field experiments and found to be in general agreement with those reported by Butler [Conference on Binaural and Spatial Hearing, 1993]. Results of this experiment are compared with results obtained in the more typical lateralization paradigms.
Localization suppression and echo suppression aspects of the precedence effect
Authors: Yang Xuefeng , Grantham D. Wesley
Published: October 2005

The precedence effect is a phenomenon that may occur when a sound from one direction (the lead) is followed within a few milliseconds by the same or a similar sound from another direction (the lag, or the echo). Typically the lag sound is not heard as a separate event and the lag sound cannot be localized. Traditionally these two aspects of precedence (echo suppression and localization suppression) have been assumed to represent the operation of a single mechanism. The hypothesis is proposed in this study that localization suppression and echo suppression are at least partially independent phenomena. A series of experiments was conducted to test this hypothesis as well as to investigate the acoustic parameters that affect the magnitude of localization suppression and echo suppression. In the first experiment subjects showed a significantly greater degree of echo suppression than of localization suppression when presented with an extended train of lead-lag noise burst pairs. In other experiments it was shown that the magnitude of localization suppression is affected primarily by the degree of spectral overlap of lead and lag sounds, and to a lesser extent by the relative ‘‘localization strength’’ of the two sounds [P. L. Divenyi, 1078–1084 (1992)]. [Work supported by NIDCD.]
Interference with interaural time difference and intensity discrimination: Empirical and modeling study of effects of component level, spacing, and perceptual grouping
Authors: Woods William S.
Published: October 2005

The discrimination of the ITD of a 600-Hz tone was measured in the presence of other tones, with levels of 75 and 45 dB SPL/tone, spacings of 200 and 400 Hz, and in the presence of noisebands. Noiseband levels were such that excitation patterns with tonal and noise interferers were equal in a region around the target. Tonal interferers were presented with simultaneous onsets and offsets, 250-ms onset asynchrony, or continuously. Noisebands were presented continuously. At the lowest presentation level with the widest spacing and continuous tonal interferers, jnds showed no increase over the target-alone jnd, i.e., no interference. Decreased spacing or increased level caused increased jnds (i.e., interference). Lesser-experienced subjects showed more interference in the asynchronous-onset condition than in the continuous-tone condition. Measurements of intensity jnds in a subset of conditions yielded similar results. These results are consistent with interference caused by spread of excitation and pulsing of interferers being as significant as interference expected from auditory grouping. A filter-based cross-correlation model for ITD processing is shown to be consistent with past and present ITD jnd results. [Work supported by NIDCD (Grant DC00100).]
Adaptive control of sound reflection
Authors: Beyene Samson , Burdisso Ricardo
Published: October 2005

The presence of an air cavity behind a layer of absorptive material creates an impedance-release boundary condition at the interface between the material and the air cavity. This boundary condition results in high absorption coefficients over relatively narrow frequency bands. These high absorption values depend on the depth of the air cavity which in turn is a function of frequency. Therefore, the impedance-release boundary condition occurs only at a few discrete frequencies. The active control system uses an adaptive feed-forward filtered-x LMS control algorithm to maintain the impedance-release boundary condition over a broad frequency range. This is done by separating the incident and reflected vector intensity within the air cavity using two microphones and a wave deconvolution circuit. A speaker inside the cavity is used as an actuator to minimize the reflected intensity vector. The result is optimum absorption over a broad frequency range, including low frequencies. Very good agreement is found between numerical simulation and experimental results. The experimental results show high absorption coefficient (0.8–1.0) over a wide frequency band (100–2000 Hz).
Optimum design for feedforward active structural acoustic control of complex structures
Authors: Rodríguez Héctor M. , Burdisso Ricardo A. , Fuller Chris R.
Published: October 2005

An efficient design formulation for feedforward active structural acoustic control systems for complex structures and disturbances is presented. The approach consists in a multilevel optimization procedure. The upper level part is carried out in the modal domain where the optimum modal control forces and modal error sensor components which minimize the total radiated power are obtained. These optimum model parameters are then used in the lower level optimization to obtain the physical characteristics of the transducers to be implemented. This separation between the modal domain and the physical domain during the design process allows the designer to investigate different control system configurations in the upper level part with minimum computational effort. The formulation is capable to handle the design of multiple-inputs, multiple-outputs control systems and multiple frequency excitations. The proposed approach is demonstrated for the case of a simply supported plate excited by a periodic point force. The first eight modes are included in the analysis. Several different control system configurations are investigated. The results show that the optimum configurations yield reductions in the total radiated power of up to 26 DB, depending in the number of control channels and the amount of allowed control effort. [Work supported by the Office of Naval Research.]
Anomalous low-pitched tones from a bowed violin string
Authors: Hanson Roger J. , Halgedahl Frederick W. , Guettler Knut
Published: October 2005

With a bow force greater than the Schelleng maximum and careful control, it will be demonstrated that it is possible to produce sounds on a violin of definite pitch ranging from approximately a musical third to a twelfth or more below the normal pitch. The lowered pitch is in agreement with the fundamental frequency of the observed harmonic series. The fundamental itself is very weak if the sounds are produced on the open G string. Mari Kimura has utilized the effect in performances [NewYorkTimes, 21 April 1994, p. B3, and Strings, Sept./Oct. 1994, 60–66]. These anomalous low frequencies (ALF) occur when the bow force is great enough to prevent the Helmholtz kink from triggering the normal release of the string from the bow hair. As a result of pronounced bow-nut and bow-bridge reflections there is at the bow a very complex string waveform, some portion of which regularly triggers the slipping of the string. ALF can also be produced on a bowed string mounted on a steel beam, where the motion is detected optically. Computer simulation is used to show how a string can be forced to vibrate at frequencies lower than the natural fundamental frequency of the string.
‘‘Subharmonics:’’ An extended technique for the violin
Authors: Kimura Mari
Published: October 2005

It has been found that unusual ways of drawing the bow on the violin could produce pitches that are as low as one octave below the fundamentals. The technique was developed from a bowing exercise meant to improve sound production by scratching harder on the string. After much practicing, all the audible transient noises were eliminated including the fundamentals, so that only those lower notes could be heard. Then the sounds were polished into steady, clear, and loud ‘‘subharmonic’’ pitches, suitable for musical purposes. It was not until April 1994, that these ‘‘subharmonics’’ were publicly introduced as a musical element at Kimura’s New York debut recital in the third movement of ALT for solo violin by Kimura, which will be performed during this presentation. In ALT, ‘‘subharmonics’’ were incorporated into the music, requiring precise control of the bow in order to freely switch between the new technique and the usual bowing. The musical effect was noted as a ‘‘revolutionary bowing technique... astonishing effect’’ [E. Rothstein, TheNewYorkTimes (21 April 1994)]. It was also found that seconds and thirds below the fundamentals can be produced in a similar manner [R. Neuwirth, Strings, Sept./Oct. (No. 44, 1994)].
The effect of extraneous stimuli on sample discrimination for frequency
Authors: Neff Donna L. , Kessler Christina J.
Published: October 2005

Four listeners completed 2IFC sample-discrimination tasks for frequency (SDF), in which they judged which of two tones or tone pairs was drawn from the higher of two frequency distributions. The distributions were separated in mean frequency by 200 Hz (100-Hz s.d.). The main conditions examined SDF at 2100/2300 Hz, in isolation or with random-frequency tones (Gaussian distributed, 100 Hz s.d.), fixed-frequency tones, or noise bands (600-Hz wide) added on both sides of the SDF region, or a single random-frequency tone added on the high or low side. Stimuli were equated in total power. Frequency position of the extraneous stimuli relative to the SDF region was varied. Flanking random-frequency tones produced the largest decrements in performance, with most listeners showing little recovery even at maximum distance from the SDF region. A single random-frequency tone below the SDF region degraded performance more than one presented above. Flanking fixed-frequency tones also produced large performance decrements that were relatively independent of distance. Noise bands produced little interference unless they overlapped the SDF region. Thus SDF can be affected by distant extraneous tonal stimuli, particularly those sharing similar patterns of variation. [Work supported by NIDCD.]
Modeling the perceived urgency of multitone signals
Authors: Haas Ellen C. , Edworthy Judy
Published: October 2005

In some environments, there is a serious mismatch between the perceived (psychoacoustic) urgency of a warning and its situational urgency (the urgency associated with the situation or condition that the signal represents). Several parameters which were believed to affect the perceived urgency of multitone auditory warning signals were investigated in a factorial experiment. These parameters were fundamental frequency (200, 500, and 800 Hz), time between signals (0, 300, and 500 ms), and signal level (5, 25, and 40 dB Lin SPL above the ambient noise level of a sound-treated chamber). The stimuli were 27 auditory signals, each signal consisting of a train of four repetitions. Free-modulus magnitude estimation quantified the relationship between auditory signal parameters and changes in perceived urgency. The effect of the three variables on perceived urgency will be discussed. The results of this experiment will be used in the development of a mathematical model in which the relationship between signal parameters and perceived urgency can be defined.
IWAIF model predictions of a profile analysis task
Authors: Ericson Mark A. , Feth Lawrence L. , Krishnamurthy Ashok K.
Published: October 2005

A profile analysis task was chosen to test the predictions of a multichannel IWAIF model. A 2Q, 2AFC three up-one down, adaptive staircase procedure was incorporated with a 20-dB roving signal level to determine the detection threshold of a 1-kHz target tone in each profile. The masker profiles included 11 adjacent of the 21 possible components ranging from 200 to 5012 Hz. The components of the profile were spaced logarithmically at frequency ratios of 1:1.18 so that no two components fell into any one critical band. The fixed frequency target tone was placed at either the second lowest, middle, or second highest component in each 11 component profile. The IWAIF model predictions are compared with channel theory predictions [Durlach etal., 63–72 (1986)] for listener performance in the profile analysis task. [Work supported by AFOSR.]
Discrimination of static versus dynamic, and log versus harmonic profiles
Authors: Watson Charles S. , Drennan Ward R.
Published: October 2005

‘‘Profile’’ stimuli consisting of multiple simultaneous fixed-frequency sinusoidal components are more representative of naturally occurring sounds than the spectrally simpler waveforms more often used in psychoacoustic experiments. However, most naturally occurring sounds are characterized by dynamic rather than static spectra, and by harmonically spaced rather than the log-spaced components used in most profile experiments. In preparing to study a variety of dynamic profiles, the discriminability of static and frequency-glide profiles was determined, using both log- and harmonically spaced components. Discriminations were based on the detection of an intensity increment added to the mid-frequency component of 11-component, 400-ms profiles. Each profile had a starting frequency range of 200 to 2000 Hz. Dynamic profiles increased in frequency continuously over their 400-ms durations. Each subject was run under four stimulus conditions (static-log, static-harmonic, dynamic-log, dynamic-harmonic) generally for a minimum of 2000 trials per condition, in an adaptive-tracking procedure. Mean differences between asymptotic thresholds for the stimulus conditions were small compared to differences among the subjects. Harmonic yielded somewhat lower thresholds than the log-spaced components, while very modest differences, if any, were found between static and dynamic profiles. [Work supported by grants from NIH/NIDCD and AFOSR.]
Stimulus-driven, time-varying weights for comodulation masking release
Authors: Buus So/ren , Zhang Lei Ji , Florentine Mary
Published: October 2005

This study tests the hypothesis that CMR is mediated by ‘‘listening in the valleys’’ [S. Buus, 1958–1965 (1985)]. Detectability was measured for signals consisting of 6 consecutive 25-ms, 1-kHz tone bursts presented in a 50-Hz wide masker or in maskers consisting of seven 50-Hz wide noises, one critical band apart, with either correlated or uncorrelated envelopes. Each burst varied randomly around masked threshold according to Gaussian distributions with 3- or 6-dB standard deviations. For each listener and condition, the responses from 5000 trials were sorted to construct conditional psychometric functions for d’ as a function of burst energy for 10 ranges of short-term level of the on-frequency masker band during the burst. The slopes of these functions for three normal listeners decrease markedly with increasing short-term masker level for the correlated multiband masker, but are largely constant for the other maskers. This indicates that the weight applied to the signal channel is high when the masker level is low and viceversa for the correlated masker, but is approximately constant for single-band and uncorrelated mutliband maskers. This finding provides direct evidence that CMR is mediated by ‘‘listening in the valleys.’’ [Work supported by NIH-NIDCD R01DC00187.]
Gating effects in CMR
Authors: Hall J. W. , Grose J. H. , Hatch D. R.
Published: October 2005

CMR is sometimes smaller for gated than continuous maskers. Our current work on CMR for multiple, comodulated narrow noise bands indicates that the gating effect is minimized when the number of comodulated bands is large. The present study extends this work to the CMR paradigm where pure tone signal thresholds are obtained as function of modulated and unmodulated noise bandwidth. Results indicate that in unmodulated noise, gated and continuous thresholds do not differ, regardless of noise bandwidth. In modulated noise, thresholds are often higher for gated noise than continuous noise when noise bandwidth is narrow, but not when the noise bandwidth is considerably wider than the auditory filter bandwidth. These results may suggest that gating effects on CMR are generally small when comodulation information is available across a large number of auditory channels. Results will be discussed in terms of competing cues for auditory grouping. [Research supported by AFOSR.]
Roughness and the critical bandwidth at low frequency
Authors: Lin Jian-Yu , Hartmann William Morris
Published: October 2005

A plausible model represents the sensation of roughness as the product of three factors, a speed factor that increases with increasing modulation rate, a temporal modulation transfer function (TMTF), and a spectral window that is determined by auditory filter widths. At high signal frequency, where critical bands are broad, the TMTF is the controlling factor. At low signal frequency the auditory filters dominate. Therefore, roughness experiments can serve to measure the critical bandwidth at low frequency. To implement this idea, listeners adjusted both the modulation rate for AM signals and the beat rate of beating sine pairs in order to achieve maximum roughness. The center frequency varied from 2000 to 70 Hz, and SPLs were 60 and 80 dB. The data could be fitted with a model in which roughness is summed over all auditory filters. The data require the critical bandwidth parameter to continue to decrease with decreasing frequency below 500 Hz so that it becomes considerably narrower than critical bands from the bark scale. [Work supported by the NIDCD.]
On simulation of sonic-boom propagation with realistic modeling of the atmospheric turbulence
Authors: Pierce Allan D.
Published: October 2005

The propagation of sonic booms is from high above the ground to ground level, and over the path the dominant acoustic length scales in the signatures fall within what is known as the inertial subrange, for which there is a considerable body of basic theoretical knowledge. Any model of boom–turbulence interaction that requires a value for an outer scale has very weak theoretical basis. However, the range of turbulence scales within the inertial subrange is huge and different scales have different effects. The present paper argues that the turbulence can be split in a well-defined and logical manner, with the resulting scale at which the split occurs serving as an outer scale with respect to the turbulence that affects the rise times of sonic booms. Although molecular relaxation accounts for a substantial fraction of the rise times, the bulk of the thickening during daytime overflights is associated with turbulence. A theory for such thickening that simultaneously took both nonlinear steepening and turbulence into account had been proposed in the early 1970s by Plotkin and George, but had been criticized for its dependence on the choice of an outer scale. The proposed partitioning removes this objection and combines the salient ideas of Plotkin and George with others proposed by the author during the same epoch for the effects of inertial-subrange-scale turbulence on rise times.
Achieving high quality speech at low bit rates: Fundamental issues in psychophysics
Authors: Atal Bishnu
Published: October 2005

Excellent progress has been made in recent years in synthesizing high quality natural sounding speech at bit rates as low as 8 kb/s. Rapidly increasing demand for speech transmission over digital radio channels requires that the bit rates be decreased even further—to 4 kb/s or lower. But, one is not able to achieve high quality speech at these low bit rates. Why? Low bit rate speech coders exploit auditory masking to mask the quantizing noise in the coded signal. The current models of auditory masking are valid only for relatively stationary segments, such as vowels, in the speech signal. Much of the distortions we heard in low bit rate coders are introduced during transitional segments, resulting from lack of psychophysical data on perception of such distortions. A human listener is able to detect such distortions in the coded version without comparing the coded version with the original. Why is it so? How does our hearing system know that the coded signal is distorted? Such questions must be answered before progress can be made in achieving high quality speech at low bit rates.
On the perceptual distance between speech segments
Authors: Ghitza Oded , Sondi M. Mohan
Published: October 2005

For many tasks in speech signal processing it is of interest to develop an objective measure that correlates well with the perceptual distance between speech segments. (Speech segments means pieces of a speech signal, of duration 50–150 ms.) Such a distance metric would be useful for speech coders at low bit rates because perturbations introduced by such coders typically last for several tens of milliseconds. It would also be useful for automatic speech recognition in adverse conditions. Since human beings perform well in spite of gross distortions of the signal (e.g., due to reverberation, noisy environments, etc.) it is justifiable to assume that mimicking human behavior will improve recognition performance. In this talk, attempts at defining such a metric will be described. The problem is approached in the framework of the Diagnostic Rhyme Test [DRT]. The errors made by subjects were measured when judiciously chosen time-frequency ‘‘tiles’’ were interchanged between the words in each pair of the DRT test [Ghitza, 2507–2515 (1993)]. Next the same task is performed with an array of automatic speech recognizers [Ghitza and Sondhi, Comp. Speech Lang. 7(2), 101–120 (1993)], using a parametrized distance metric. Finally, the parameters of the distance metric are optimized so as to minimize the difference between the error patterns of the human listeners and those of the automatic speech recognizers.
The effects of room acoustics on normal-hearing children: Implications for intervention
Authors: Crandell Carl C. , Smaldino Joseph
Published: October 2005

It is well recognized that the acoustical environment in a classroom is an important variable in the psycho educational achievement of hearing-impaired children. To date, however, there remains a paucity of information concerning the importance of classroom acoustics for populations of children with normal-hearing sensitivity. The present discussion will examine: (1) commonly reported levels of classroom noise and reverberation; (2) the potential effects of classroom acoustics on the speech recognition of several populations of pediatric listeners (young children, children with language/articulation disorders, non-native English children, children with minimal hearing loss, developmentally disordered children, children with central auditory processing disorder); (3) intervention strategies, such as sound field amplification systems, to benefit normal-hearing children in the classroom setting; and (4) implications of appropriate classroom acoustics to the American with Disabilities Act (ADA).
Proposed guidelines for quiet areas in restaurants for hearing-impaired individuals
Authors: Moulder Ron , Lubman David
Published: October 2005

The predominant problem of the hearing-impaired individual is speech communication. In noisy and reverberant restaurants where speech communication is difficult to impossible, these individuals cannot enjoy the experience of dining out. This fact was conveyed to the US government during hearings for the Americans with Disabilities Act. A program was funded by the federal government to develop recommendations for improving the acoustical quality of restaurants so that hearing-impaired individuals could enjoy the experience of dining out. A three-phase program consisting of a literature search, the evaluation of the acoustical environment in selected restaurants, and the development of proposed recommendations to improve speech communication by hearing-impaired individuals in restaurants was carried out by Battelle. An advisory panel of professional and lay individuals was established to review the work of this program. The proposed recommendations developed focused on reducing reverberation times and noise levels in restaurants. Several recommendations are presented for achieving these goals.
Development of acoustical criteria in a school for children with hearing disabilities
Authors: Erdreich John , Herbert R. Kring
Published: October 2005

Classrooms for the teaching and rehabilitation of hearing-impaired children pose special problems in acoustical design. Both Elliot [L. L. Elliot, J. Acoust. Soc. Am. 66, 12–21; 651–653 (1979)] and Mills [J. Acoust. Soc. Am. 58, 767–779 (1975)] have shown that the levels of noise which interfere with speech perception in children without hearing impairment are lower than those which interfere with intelligibility in adults. It is logical to conclude, therefore, that the design of rooms in schools for the hearing impaired should include criteria more stringent than those for other classrooms. This approach was adopted in establishing criteria for the Summit Speech School which conducts two distinct programs of speech training for hearing impaired children: One for children from birth to age three and the other, for ages 21/2 to 5. Several factors were identified for particular attention including control of activity noise without excessive sound absorption and control of flutter echo. Among the criteria recommended were minimum sound isolation between classrooms and other spaces of STC-48 and maximum background sound levels of RC-25. Criteria for other spaces and their basis will be presented.
Internal-tide inversions from a 1000-km Pacific acoustic propagation experiment in the Pacific
Authors: Bracher Charles , Flatte Stanley M.
Published: October 2005

The Slice89 experiment sent sound pulses through the Pacific over a 1000-km path. Different identified wavefronts showed clear M2 internal-tidal signals in their arrival-time series. we have examined internal-tide models composed of the first three vertical modes in order to fit the Slice89 data. The horizontal directions of the three modes were allowed to be different. The horizontal structure of the models varied from plane waves to waves that are localized along the direction of acoustic propagation. It has been found that the best-fit internal-tide model within the parametrization, and the resulting fit is reasonable. These results along with those of other experiments in the same geographical area have been compared.
1994 moving ship tomography experiment in the western Mediterranean
Authors: Mikhin Dmitry Yu. , Aleynik Dmitry L. , Burenkov Sergey V. , Chepurin Yury A. , Selivanov Viktor G. , Molchanov Sergey Ya. , Godin Oleg A.
Published: October 2005

A tomographic network of seven moored transceivers was deployed in the western Mediterranean basin in early 1994 by IfM (Kiel, Germany), IFREMER (Brest, France), and WHOI (Woods Hole, USA) in the framework of the THETIS-2 project. Signals emitted by the transceivers were also utilized in the 8-week moving ship tomography experiment, MOST. Acoustic measurements were made with a single hydrophone and an eight-element vertical array deployed from a drifting vessel and were complemented by CTD casts. The set of observation points was chosen to provide detailed CTD and acoustic data along existing propagation paths of the THETIS-2 network for comparison of traditional and moving ship tomography under well-controlled environmental conditions as well as to create numerous additional propagation paths to improve resolution in the horizontal plane. By precise compensating the Doppler shift due to vessel drift, the data processing technique used made it possible to resolve eigenray arrivals up to 400–600 km range for different sources. Preliminary results of the MOST data processing, analysis, and interpretation will be presented at the meeting. [Work supported by ISF and INTAS.] a)On leave from P. P. Shirshov Oceanography Institute, Moscow.
On acoustic tomography of internal waves
Authors: Colosi John A. , Flatte Stanley M.
Published: October 2005

With the planned acoustic instrumentation of several of the worlds ocean basins, the possibility exists for measuring the strength of the internal wave field as a function of depth, time, and geographical position. The variability of internal-wave energies and spectra over these parameters will provide information on the sources and sinks of internal waves. In the early eighties Flatte formulated an analytic, ray-based framework for internal-wave tomography. Analysis of the SLICE89 experiment combined with supercomputer numerical simulations have recently shown that internal waves have even stronger effects on signals transmitted over 1000 km than were expected from the early ray-based theory plus the Garrett–Munk (GM) spectrum. It was found that the ray-based theory is adequate for identifiable wavefronts that turn at depths below 100 m; energy that reaches very shallow depths and energy that stays near the sound-channel axis do not behave so simply. The numerical simulations establish an internal-wave strength below 100 m of 0.5±0.25 the reference GM level, and point to known limitations of the GM spectrum in the upper ocean. Simulations to 3000 km have quantified the range dependence of internal-wave effects, and extensions of ray-based theory attempt to explain these effects.
Internal wave effects on shallow water acoustic tomography
Authors: Tang Xin , Tappert F. D.
Published: October 2005

Full-wave numerical simulations of internal wave scattering of sound in shallow water are performed with an efficient broadband PE model that also includes scattering by bathymetric variations. Bathymetric scattering by itself gives an exponentially large number of unresolvable multipaths as predicted by ray chaos theory. When internal waves that evolve dynamically in geophysical time are included, the unresolvable multipaths are found to be temporally unstable with coherence times less than a few minutes at ranges greater than a few tens of water depths in shallow water, except possibly for rare paths that are shielded from internal waves such as within strong surface ducts. Since traditional travel-time ocean acoustic tomography requires many stable, resolvable and identifiable paths, conditions that are not satisfied in shallow water at longer ranges, some other inversion method will have to be developed for remote sensing of shallow water environments. [Work supported by ONR.]
Surface suspended acoustic receiver for acoustic tomography
Authors: Spiesberger John S. , Ackerman Carter , Einfalt Bruce
Published: October 2005

A surface suspended acoustic receiver (SSAR) is shown capable of processing and transmitting acoustic tomography data through ARGOS to laboratories in near-real time. Ray theory is shown adequate for validating the data from the SSAR. [Work supported by the Strategic Environmental Research and Development Program, managed by the Advanced Research Projects Agency.]
ATOC-FACT arrival-time differences
Authors: Palmer David , Krige Leon , Brundrit Geoff , Metzger Kurt
Published: October 2005

The acoustic thermometry of ocean climate-feasibility ascension Cape Town (ATOC-FACT) propagation experiment took place during November and December of 1992. Explosive charges were detonated near the oceanic sound-channel axis at five locations in the South Atlantic off Cape Town. The resulting signals were recorded on bottom-mounted hydrophones at Ascension, Navy horizontal arrays off Bermuda, bottom-mounted hydrophones off Kaneohe and Wake and from a hydrophone array deployed from a research ship off Southern Tasmania. The primary goal of the experiment was to determine the extent that the acoustic path between Ascension and Cape Town is blocked by Bonaparte’s Seamount and other bathymetric features. Secondary goals were to determine the horizontal arrival structure at Bermuda and the temporal arrival structure at Ascension, Hawaii, and off Tasmania. After giving an overview of the experiment, we use recorded data to discuss the question of how precisely arrival-time differences of explosive signals can be measured. Detonations separated both temporally and spatially are considered. It is hoped this work will contribute to an understanding of the role explosive signals can play in the design of a global ATOC network.
Observing the ocean in the 1990s: Where do we stand now?
Authors: Wunsch Carl
Published: October 2005

Thirteen years ago, a paper with almost the same name as this present session was published by Munk and Wunsch. Ways in which oceanographers were likely to be observing the ocean 10 years later was speculated upon. The special focus was climate-scale problems with acoustic tomography and satellite altimetry as the natural complementary observational technologies. At the time, the tomographic idea was only about 4 years old, and now that it is half-way through the 1990s, it is interesting and perhaps useful to look back to understand the extent to which the vision of the role of tomography in oceanography has come to pass. A number of problems will be considered, from the technical to the sociological and attempt to make yet another forecast of where tomography in oceanography might be in another 10 years.
Silencers: The impact of active control
Authors: Eriksson L. J.
Published: October 2005

Over the past decade, powerful new adaptive techniques have been developed for the active control of sound and vibration. One of the areas that has received the most attention is the application of active control to enhance the performance of silencers used on ducts or pipes. Active silencing is often combined with passive silencing to create a new form of hybrid silencer. These silencers can be used as intake or discharge silencers for heating, ventilating, and air conditioning (HVAC) fans, industrial fans, rotary blowers, compressors, internal combustion engines, gas turbines, pumps, and related equipment. This paper will review the use of active control for these applications with particular emphasis on the different reasons generating the interest in active control for each application. Among the potential advantages of active control for silencers are improved low frequency performance, reduced flow restriction, and decreased size and cost. The degree to which these potential advantages can be realized in specific applications can vary considerably as well as the ultimate impact of the improvement for the end user. These issues will be discussed using a number of actual examples and implications for the future use of active control will be suggested.
Fiber-optic microphone based on combination of Fabry–Perot interferometry and intensity modulation
Authors: Zhou Chonghua , Letcher Stephen V. , Shukla Arun
Published: October 2005

A fiber-optic microphone based on a combination of Fabry–Perot interferometry and intensity modulation is described. The combination technique eliminates fringe counting ambiguity and provides a wide range measurement of optical phase change without any complicated electromechanical devices. The experimental data are compared with the result of a theoretical analysis. The microphone has a flat frequency response from 20 Hz to 10 kHz with a high sensitivity of 1.6 rad/Pa. The dynamic range is more than 65 dB. The frequency response and sensitivity can be modulated by changing the size and surface tension of the membrane. The microphone can be easily implemented and can be made very small. [Work supported by NSF.]