bluuebunny/crossref_metadata_2025 huggingface dataset:
A fast and accurate bistatic bottom reverberation model
Authors: Haller D. R.
Published: October 2005

A simple bistatic reverberation model has been developed to model low-frequency, wideband, bottom, or surface reverberation. The model allows for arbitrary separation of source and receiver in both range and depth, and accommodates both source and receiver beam patterns. Total reverberation is computed by the incoherent summation of the incremental contributions from each three-dimensional ray path connecting the source, scattering element, and receiver. Simplified ray path calculations are made with either a constant or a linear sound speed profile, assuming spherical spreading along each ray, and specular reflection from the flat, horizontal boundary interfaces. Acoustic characteristics of the surface and bottom are assumed to be completely represented by the reflection loss and scattering strength functions. A realistic representation of generalized bistatic scattering [D. D. Ellis and D. R. Haller, J. Acoust. Soc. Am. Suppl. 182, S124 (1987)] has been successfully used to model reverberation from explosive sound sources in the Tufts Abyssal Plain of the Northeast Pacific Ocean. Predictions from the simple reverberation model were found not only to be very similar to those of the bistatic generic sonar model but also to require much less computation time.
Similarity of a Fourier transform generalization of the Earnshaw solution for planar waves to an interacting wave model for finite amplitude effects in sound beams
Authors: Ginsberg J. H.
Published: October 2005

The Earnshaw solution for a finite amplitude planar wave, which displays amplitude dispersion, is valid for arbitrary excitation f(t) on a boundary. The form that results for small acoustic Mach numbers when f(t) is represented as the inverse of its Fourier transform F(ω) may be considered to be a coordinate straining of the linearized signal, in which the transformation has the appearance of a Fredholm integral equation in the frequency domain. In the case of an acoustic planar wave, the phase speed of all frequency components is the same, independent of frequency. A generalization of that result is obtained if one considers the possibility that the cumulative growth effect is an arbitrary function of distance. Rather than being an abstraction, this form is shown to be analogous to an earlier solution for finite amplitude sound beams [J. H. Ginsberg, H. C. Miao, and M. A. Foda, J. Acoust. Soc. Am. Suppl. 1 81, S25 (1987)]. The signal in that analysis was represented in terms of a spectrum of transverse wavenumbers, rather than frequencies, and the signal for each wavenumber was formed from interacting quasiconical waves, rather than a single planar wave. Nevertheless, the two problems share a common mathematical structure. A discussion of numerical algorithms for such models leads to some surprising observations regarding the implicit functional form of the Earnshaw solution. [Work supported by ONR.]
Scattering of sound by sound from two Gaussian beams
Authors: Darvennes Corinne M. , Hamilton Mark F.
Published: October 2005

The scattering of sound by sound from Gaussian beams that interact at small angles is investigated theoretically with a quasilinear solution of the Khokhlov-Zabolotskaya nonlinear parabolic wave equation. The analytical solution, which is valid throughout the entire paraxial field, is a generalization of the result obtained for parametric receiving arrays by Hamilton, Naze Tjøtta, and Tjøtta [J. Acoust. Soc. Am. 82, 311–318 (1987)]. Significant levels of scattered difference frequency sound are shown to exist outside the nonlinear interaction region. An asymptotic formula reveals that difference frequency sound is scattered in the approximate direction of k1-k2, where ki is the wave vector associated with the direction and frequency of the ith primary beam. Computed propagation curves and beam patterns demonstrate the dependence of the scattered radiation on source separation, frequency ratio, interaction angle, and focusing. Results are also presented for the scattered sum frequency sound. Comparisons are made with the general asymptotic results presented by Berntsen, Naze Tjøtta, and Tjøtta [J. Acoust. Soc. Am. Suppl. 1 83, S4 (1988)], which are valid for arbitrary interaction angles, source separations, and amplitude distributions. [Work supported by ONR.]
Directing attention in pitch and time: Effects on the perception of hidden melodies
Authors: Dowling W. Jay
Published: October 2005

Listeners attended to a target tone in the middle of a familiar melody that was interleaved with distractor notes. The target occurred in either expected or unexpected locations in pitch or in time. Expectancy was controlled by means of a cue melody preceding the embedded target. In half the sessions, the cue indicated the pitch level of the target, and listeners judged the time at which it occurred. In the other sessions, the cue indicated time, and listeners judged pitch. The cue was 83% valid, and expected targets matched the cue while unexpected targets did not. Rate of presentation was adjusted within each session to produce approximately 80% accuracy. Temporal expectancies had a much greater effect on reaction times (RTs) than did pitch expectancies. RTs for pitch judgments were about 130 ms longer when targets occurred at unexpected (versus expected) times, while RTs for time judgments were only about 25 ms longer for targets at unexpected (versus expected) pitches. Rhythmic organization provides a basis for the listener to direct attention to critical events in a rapid auditory stream.
The subjective size of musical intervals
Authors: Plamondon Lyne
Published: October 2005

The musical scale and the psychophysical scales of pitch appear to be in contradiction with one another. The debate centers on whether or not pitch is a log function of frequency. Previous results on the listeners' judgment of ratios and differences in pitch between pairs of tones agree with the musical scale [R. Elmasian and M. H. Birnbaum, Percept. Psychophys. 36, 531–537 (1984)]. In the present experiment, the subjective size of musical and nonmusical intervals was measured by magnitude estimation and category rating, using subjects without musical training. Pure tones as well as complex tones (harmonic series) were used. Frequencies ranged from 110–2145 Hz. For each pair of tones presented melodically, subjects were instructed to rate the pitch of the second tone relative to the pitch of the first one. Results show that subjects use the same comparison operation for both ratio and difference judgments of pitch. The effects of frequency range, spectral composition, size of interval, and musicality of the ratios on the subjective size of musical intervals are discussed. [Research supported by NIH.]
The perception of time-separation pitch by dolphins
Authors: Au Whitlow W. L. , Pawloski Jeffrey
Published: October 2005

A dolphin was required to discriminate between rippled and nonrippled noise projected by an underwater transducer. Random noise was summed with its delayed replica to produce noise having ripples separated by 1/T Hz in the frequency domain, where T is the delay time. The dolphin detected the cos − rippled stimulus at a correct response level of at least 75% for delays between 15 and 500μs, and the cos + rippled stimulus for delays of 13 to 190 μs. The dolphin's sensitivity to rippled noise was measured by attenuating the delayed replica for different delays. The dolphin was most sensitive for a delay of 100 μs. Its sensitivity at 100 μs was 5 dB better for the cos + than the cos − stimuli. The broadband cos + noise was also filtered in different 1/3 octave bands to determined if the animal's sensitivity to rippled noise was a function of the center frequency of the noise. The dolphin's performance was relatively constant as a function of center frequency. The overall results suggest that dolphins may be able to perceive time-separation pitch.
Differential frequency sensitivity of humans and monkeys for tone versus vowel stimuli
Authors: Sinnott Joan M.
Published: October 2005

Humans and monkeys (Macaca fuscata) were compared in their abilities to discriminate frequency changes (a) in a 2-kHz pure tone and (b) along a steady-state /I/ to /i/ vowel continuum (F1 = 390 − 270 Hz; F2 = 1990–2290 Hz; F3 = 2550–3010 Hz). A repeating standard AX procedure was used and stimuli were presented monaurally through the right car. For the tone stimulus, human and monkey DLs were markedly divergent; at low SLs, the human DL (6 Hz) was about 1/5 that of monkeys (30 Hz) while, at high SLs, the human DL (4 Hz) decreased to about 1/20 that of monkeys (80 Hz), as monkeys became less sensitive. For the vowel stimulus, human and monkey DLs (measured relative to changes in F2) were more similar: The human DL (15 Hz) was about 1/3 that of monkeys (45 Hz), and SL functions were parallel and flat for both species. Results indicate that monkeys provide a much better (although not perfect) model of human sensitivity for a complex vowel stimulus than for a simple tone stimulus. [Work supported by NIH.]
Speech perception by budgerigars (Melopsittacus undulatus): Spoken vowels
Authors: Brown Susan D. , Dooling Robert J. , Okanoya Kazuo
Published: October 2005

Budgerigars (parakeets) were trained on a same/different task using operant conditioning. Response latencies were used to construct similarity matrices, and multidimensional scaling procedures were then used to produce spatial maps of these speech stimuli. Budgerigars were tested on stimulus sets consisting of natural exemplars of different vowels produced by a number of male and female talkers. In spite of variation among talkers, budgerigars showed evidence of perceptual categories for different vowels as well as for male and female talkers. The two-dimensional perceptual space for budgerigars discriminating among vowels bears a striking resemblance to the familiar F1-F2 vowel space described by Peterson and Barney [J. Acoust. Soc. Am. 24, 175–184 (1952)]. This suggests that budgerigars use first and second formant information in discriminating among spoken, sustained vowels. [Work supported by NIH.]
A study on the speaker-independent feature extractinn of Japanese vowels by neural networks
Authors: Irino Toshio , Kawahara Hideki
Published: October 2005

The feature extraction characteristics of three-layer back propagation neural networks, when applied to speaker-independent vowel/gender recognition tasks, are investigated. Speech samples are monosyllabic vowels extracted from a syllablic data base of 100 Japanese syllables digitized at a sampling rate of 16 kHz spoken by 100 different speakers. Several LPC-based parameters and physiology-based parameters are tested for input representations. The results reveal that three hidden units are necessary to discriminate five Japanese vowels. Close investigation of hidden unit functions reveals that distributed representations of the targets are developed as hidden unit activation patterns. The frequency domain interpretation of the input weights of hidden units, using autocorrelation inputs, shows that the network extracts conventional knowledge on vowel formant structure. The relations between hidden unit activation patterns and descriptive features, as well as their implications to speech perception research, are also discussed.
Recoguition of vowels using artificial neural networks
Authors: Leung Hong C. , Zue Victor W.
Published: October 2005

This paper is concerned with the application of artificial neural networks to phonetic recognition. This work is motivated by the observation that improved knowledge of feature extraction is often overshadowed by relative ignorance on how to combine them into a robust decision. The goal is to investigate how the self-organizing framework of artificial neural networks can be exploited to enable different acoustic cues to interact. The investigation is couched in experiments that recognize the 16 vowels in American English, using some 10 000 tokens in all phonetic contexts. The tokens were extracted from 1000 sentences spoken by 140 males and 60 females. It was found that, by replacing the mean-squared error metric with a weighted one to train a multilayer perception, better recognition accuracy and rank order statistics were consistently obtained. Using the two-layer perceptron in a context-independent manner, a top-choice accuracy of 54% was achieved, which compares favorably with results reported in the literature. Context-dependent experiments reveal that heterogeneous sources of information can be integrated to improve recognition performance. The top-choice accuracy of 67% is comparable to the average agreement among listeners. Finally, it was found that the rate of improvement on recognition accuracy may be used as a terminating criterion for training, and that reasonable performance can be achieved using as few as 800 training tokens. [Work supported by DARPA.]
Effects of network topology on speech categorization
Authors: Merrill John , Anderson Sven , Port Robert
Published: October 2005

Recognizing speech requires the identification and anlysis of temporally distributed cues. A system must either examine a sufficiently long window at a single glance or else internally accumulate stimulus information. Sequential networks follow the second path by storing information internally in state nodes. Feed-forward networks do not maintain their history internally and thus require that the speech signal be presented in fixed windows. The performances of sequential and feed-forward networks at recognition of auditorily preprocessed stop-vowel syllables are compared. Several feed-forward networks were trained by presenting a whole syllable to the network as a single token and requiring categorization. The sequential networks were more robust within and across speakers than the feed-forward networks. Unfortunately, the use of the back-propagation algorithm to train a sequential network requires presentation of a desired output at every time slice. This forces arbitrary choices for specifying target outputs. The results suggest that techniques based on back-propagation will prove inadequate to train networks to perform speech categorization across variations in speaker. This difficulty could be obviated by employment of a learning paradigm that does not require immediate feedback, such as a stochastic learning algorithm. [Work supported by NSF.]
Using an adaptive network to recognize demisyllables in continuous speech
Authors: Kamm Candace A. , Landauer Thomas K. , Singhal Sharad
Published: October 2005

A nonlinear, multilayer associative network was trained on a speech recognition task using continuous speech. Naturally spoken 14-syllable “sentences” from one talker were preprocessed to produce a 15-band spectral representation incorporating several transformations introduced by the peripheral auditory system on acoustic signals. Input nodes to the network represented a 150-ms window through which the spectral representation passed in 2-ms steps. A single layer of 20 hidden nodes was used. Output nodes represented seven initial demisyllables whose target values were specified based on a human listener's identification of the sounds heard during the input segment. The network was trained to criterion using a variant of the back-propagation learning algorithm [Rumelhart et al., Nature 323, 533–536 (1986); Landauer et al., Proc. Cog. Sci. Soc., 531–536 (1987)]. A minimum-error-rate figure of merit (derived from signal detection theory) was used to evaluate the effect of the size of the training corpus on the network's performance. Minimum error rate on a test corpus of demisyllables spoken in different contexts decreased from 5.2% to less than 2% as the number of sentences in the training corpus was increased from one to seven.
Comparison of standard ASR front ends and auditory models in neural net-based automatic speech recognition
Authors: Terry Mark , Hermansky Hynek
Published: October 2005

Recent work [Renals and Terry, European Conference on Speech Technology, Edinburgh, United Kingdom (1987)] reported on a small multispeaker isolated digit automatic speech recognition (ASR) experiment using a backward propagation neural net system with a synchrony auditory model as its front end. In spite of fairly crude temporal normalization, the system was capable of better than 80% ASR accuracy. The question remains if the reported performance is due to (a) the paticular front end, (b) the particular neural net-based classification, (c) the particular time-normalization scheme, or (d) the combination of all factors. The goal here is to isolate these factors. In the present contribution, different front ends in the neural net-based ASR are systematically evaluated. Several standard ASR front ends are compared with the synchrony auditory model and the perceptually based linear predictive auditory model front ends in both the speaker-dependent and the speaker-independent ASR. The speed of learning and the ASR accuracy of the compared recognizers are reported and discussed.
An artificial neural network-based codebook search technique
Authors: Pope Charles , Atlas Les , Nelson Charles
Published: October 2005

An unsupervised learning artificial neural network (ANN) [T. Kohonen, Helsinki University of Technology Report TKK-F-A601 (1986)] was modified for a vector-quantized PCM (VPCM) codebook search problem. The ANN was structured such that each element of the network had associated with it a single codebook vector. The amount of processing required at each element of the network was used to derive an upper bound on the number of network iterations. Simulations were performed to determine the effect that various network parameters had on the speed of network convergence. The parameter values offering the greatest performance were applied to the network. The speed and computational complexity of the ANN solution to this problem were then compared to the same criteria for a standard linear (full-codebook) search technique. Analysis and test results indicate that the ANN approach can provide the speed of a tree search coupled with the minimum memory characteristics associated with a linear search, at the expense of requiring a multiprocessor configuration. [Work supported by NSF.]
Enhanced speaker identification through the use of cepstral coefficients and redundant time frame elimination
Authors: Thomas Timothy R. , Papcun George J. , Willie Stan , Ganesan Kalyan
Published: October 2005

A text-independent speaker identification system was developed on two male and two female speakers with similar accents. Each read a phonetically balanced list of ten sentences. The system was trained repeatedly on a rotated set of nine sentences and tested on the remaining one. Speech during successive 16-ms windowed time slices was described by 14 cepstral coefficients. Unit direction vectors were used to characterize each sentence from each speaker. A nearest-centroid, nearest-neighbor, or improved perceptron neural net training procedure was used to define decision regions. When the data were preprocessed so as to remove time slices that were similar in all speakers, discriminability was enhanced and error-less identification was obtained. The success of this system appears to result primarily from the ability of the cepstral coefficients to capture the speaker-dependent information in the higher formants and from the accentuation of this information by the preprocessor.
Comparative study of ASR front-ends in noise
Authors: Junqua Jean-Claude , Wakita Hisashi
Published: October 2005

In automatic speech recognition (ASR) of speech corrupted by noise, the performance tends to deteriorate rapidly depending on the choice of analysis method and distance measure. In order to evaluate the recognition performance for several analysis methods and distance measures, a series of isolated word recognition experiments was performed. Analysis methods selected are critical-band filtering, perceptually based linear prediction (PLP), linear prediction (LP), and time synchronous linear prediction (SLP). The weighted Euclidean distance with different weightings [unity, root power sums (RPS), and exponential filtering] was applied in the cepstrum domain. Experiments were carried out for clean speech and for two noise conditions (white and low-pass filtered white, added to the clean speech) at different SNR ratios (25 to 5 dB), using an alphanumeric vocabulary (ten speakers). It is shown that improvements in robustness of the recognizer in noise can be achieved by a proper selection of analysis method and cepstral weights used in the front-end. Improvements are found over the RPS distance measure (previously shown to be useful in noise conditions with LP and PLP analyses) [B. Hanson and H. Wakita, Proceedings ICASSP 86 (IEEE, New York, 1986), pp. 757–760] by use of the general exponential lifter.
Feature-based automatic syllable and stress detection
Authors: Williams Briony , Dalby Jonathan
Published: October 2005

The importance of syllable structure and stress level as determinants of segmental temporal and spectral variability makes automatic syllable detection and stress estimation a very desirable goal for continuous speech recognition research. In this paper, a rule-based system is described for locating syllables in continuous speech and for making a two-level stress assignment. Location of syllable nuclei and rough estimation of syllable boundaries are performed using a smoothed midfrequency “sonorant” energy contour, a frication detector, and Sonorant consonant detectors. First pass classification of detected syllables as stressed, unstressed, or uncertain is based on the relative energy levels and utterance-position-normalized durations of syllables in a three-syllable window. Fundamental frequency information is then used to reclassify uncertain cases as either stressed or unstressed. Preliminary evaluation of the system's current performance on a multispeaker data base yielded 77% correct location and classification of syllables. Although improvement is necessary, this result is encouraging.
A segmentation algorithm based on spectral variance
Authors: Kumar A. , Wakita H.
Published: October 2005

Presented is a segmentation algorithm based on spectral variance. The speech signal is first segmented into spectrally stable segments by using a median smoothed spectral variance over a 70-ms window. The segment boundaries are placed at the maxima in the spectral variance and the minima give typical frames for the segments. The spectral variance peak for the glides is generally very small because of their smooth transition. Hence, the glides are segmented by using second and third formant trajectories. Extraneous segments outside the word boundaries are eliminated by adaptive silence detector. The segments are then assigned broad phoneme classes by using a tree classifier on the LPC cepstral coefficients for the typical frames. The fricative and nonfricative segments are distinguished by the normalized differenced speech signal. The tree classifier is separately trained on hand-labeled databases. Preliminary experiments show that 90% of the segments are detected. Results for the 104-word keyboard vocabulary for six males and four females and continuous speech will be presented.
Speech recognition using a synthesized codebook
Authors: Smith L. A. , Scott B. L. , Goodman R. G. , Lin L. S. , Newell J. M.
Published: October 2005

Speech sounds generated by a simple waveform synthesizer were used to create a vector quantization codebook for use in speech recognition. Recognition was tested over the TI-20 isolated word database using a conventional DTW matching algorithm. Input speech was filtered to limit the bandwidth to 300–3300 Hz, then was passed through the Scott Instruments Coretechs process, implemented on the SI2010 signal processing chip, to create the speech representation for matching. Synthesized sounds were processed in software by an SI2010 emulation program. SI2010 emulation and recognition were performed on a DEC VAX 11/750. The original codebook contained 109 vectors. This codebook was decimated through the course of the experiments, based on the number of times each vector was used in quantizing the training data for the previous experiment. Recognition scores are presented for progressively smaller codebook sizes, as well as for the baseline condition (no vector quantization).
An efficient, robust speaker-independent algorithm
Authors: Scott Brian , Lin Lisan , Newell Mark , Smith Lloyd
Published: October 2005

The algorithms described have yielded speaker-independent scores of 95.1% on the 20-word TI database obtained from the National Bureau of Standards. Results were obtained by training the system on half of the speakers in the database, testing on the other half, and then reversing the order. Training was done with the 10 training tokens per speaker per word only. Testing was on the 16 test tokens per speaker per word. The total number of test trials was 5120. The recognizer uses conventional methods for time normalization and matching. Time normalization is linear and scoring is accomplished with a simple differencing algorithm weighted by variances. Storage requirement is 3072 bits per word. Most of the speaker normalization is accomplished by the proprietary signal processing method developed by Scott Instruments. Aside from the amplitude normalization routines, no floating point arithmetic is used. All signal processing is temporally based. The front end process can be adapted for use with dynamic time warping algorithms or feature based algorithms. The system is, therefore, extensible to connected speech.
Human and machine performance on speaker identity verification
Authors: Feustel Timothy C. , Logan Robert J. , Velius George A.
Published: October 2005

Two experiments were conducted to identify acoustic features for speaker identity verification (SIV) that are used by humans and not by cepstral-based algorithms. Although these algorithms generally out-perform human listeners for randomly selected comparisons between single-word utterances, this approach was to analyze human performance on comparisons that could not be effectively discriminated by machine. Experiment 1 showed that humans could perform at high levels of accuracy on these comparisons suggesting that either information exists that is not captured by the algorithms, or that the information is coded by the algorithms but is not used effectively. The second experiment consisted of three stimulus conditions for SIV; digitized speech signals, noise-excited resynthesized LPC signals, and error prediction signals from the LPC. Results indicated high levels of performance in the natural and error prediction signal conditions and performance near chance in the noise excited condition, thus suggesting that the error signal provides valuable information that allows humans to distinguish between speakers. It may be possible to improve verification algorithms by adapting current models to more accurately utilize information used by human listeners.
Auditory responses to the envelopes of pseudorandom noise stimuli in humans
Authors: Dobie Robert A. , Wilson Michael J.
Published: October 2005

Averaged scalp potentials evoked by continuous pseudorandom noise can be cross correlated with the evoking stimulus, yielding a cross-correlation function (CCF) that reflects neural phase locking and is quite sensitive for low-frequency stimulus components [M. J. Wilson and R. A. Dobie, Electroencephalogr. Clin. Neurophysiol. 66, 529–538 (1987)]. However, for higher frequency signals, replicable CCFs can only be obtained at moderate to high intensity. Since auditory neurons also respond to envelopes of complex sounds, even for high-frequency carriers, scalp responses were compared to the envelopes of complex sounds; the resultant envelope cross-correlation functions (ECCFs) contained replicable response components primarily below 1000 Hz, regardless of the evoking stimulus spectrum. The ECFF thresholds for three octave-band stimuli (830–1562, 1611–3125, and 3174–6201 Hz) were more sensitive than CCF thresholds (p = 0.018), averaging 35-dB spectrum level for ten normal subjects. When stimuli with only odd harmonics were used, replicable odd-component responses were seen only in the spectral range of the stimuli, while even-component responses (presumably to stimulus envelope) were seen only in the low-passed scalp responses.
The effect of stimulus repetition rate on interaural ABR latency differences in normal and hearing-impaired subjects
Authors: Hannley Maureen , Weichert Lorene , Wedekin Sarah B.
Published: October 2005

Twenty-five subjects with normal hearing and 15 with bilateral high-frequency sensorineural hearing impairment were presented click stimuli at four repetition rates (11, 33, 67, and 89 clicks/s) to access interaural symmetry of wave V latency and amplitude. Predictably, amplitude was significantly decreased and latency increased across all subjects at the higher repetition rates. In both groups, however, interaural latency differences (ILDs) remained with the 0.3-ms normal criterion previously established for slow repetition rates. Thus, both abnormal repetition rate effects and interaural latency differences can serve as reliable indicants of retrocochlear auditory dysfunction, even when peripheral hearing impairment exists.
The auditory brain-stem response in children with a history of OME
Authors: Chambers Ron D. , Rowan Lynne E. , Novak Michael A.
Published: October 2005

The auditory brain-stem response (ABR) was used to evaluate the auditory-neural status of children with a history of frequent or persistent otitis media with effusion (OME). Subjects were selected from the case-load of a board certified otolaryngologist and from the population at large. They were pair-matched by age and sex to form an otitis-prone and a control group. The otitis-prone group included only those children whose degree of OME involvement was sufficient to warrant tympanostomy tube placement. Stimuli were rarefaction clicks presented at 85 dB nHl at a rate of 21.4/s. The latency of wave III and the III-I interwave interval were longer for the otitis-prone subjects than for the control subjects, which agrees with earlier results [R. C. Folsom, B. A. Weber, and G. Thompson, Ann. Otol. Rhinol-Laryngol. 92, 249–253 (1983)]. The results and precautions in interpretation will be presented. [Work supported by Carle Research Foundation.]
Effect of internal relaxation on the optoacoustic signal in CS2
Authors: Thompson Charles , Bass Henry E. , Raspet Richard
Published: October 2005

The optoacoustic signal in CS2 has been observed using an experimental configuration similar to that of Sullivan and Tam [B. Sullivan and A. C. Tam, J. Acoust. Soc. Am. 75, 437–441 (1984)] and has been compared to theory and similar results in propanol. The comparisons illustrate that an internal relaxation process is affecting the observed waveform. Assuming a simple three-level system for internal energy and using independently measured vibration-to-translation transfer rates leads to the conclusion that the relaxation process observed is associated with electronic-to-translation or electronic-to-vibration energy transfer with a relaxation time of 90 ns. [Work supported by the Office of Naval Research.]
Acoustic imaging by a method of image projections
Authors: Rouseff Daniel , Porter Robert P. , Tsang Leung
Published: October 2005

A method of three-dimensional acoustic imaging based on image projections is derived. A weakly scattering object is illuminated by a plane wave and the scattered field is recorded. Inversion methods based on projections that operate directly on the scattered field are inaccurate due to diffraction effects. By backpropagating the scattered field onto a plane in the image region, diffraction effects are reduced and a projection of the object is generated. Conventional x-ray inversion techniques may then be used to image the object. The relationship between the backpropagated field and the projection is derived both for high-frequency incident waves and low spatial frequency scatterers. Backpropagation additionally allows the use of curved or misaligned recording surfaces. [Work supported by ONR.]
Auditory demonstrations on compact disc
Authors: Houtsma A. J. M. , Rossing T. D. , Wagenaars W. M.
Published: October 2005

In 1984, the Committee on Education in Acoustics of the ASA discussed the possibility of reissuing the “Harvard Tapes,” a series of audio demonstrations on tape cassettes originally published in 1978 by David M. Green at the Harvard Laboratory of Psychophysics. The present authors undertook the job of evaluating and updating the content of the original tapes, producing new audio material, and writing a new explanatory book. The compact disc (CD) medium was chosen to replace magnetic tape. Through substantial cooperation from Eindhoven University of Technology, Northern Illinois University, the Philips Company, and the ASA, a CD with a 92-page booklet was developed containing groups of auditory demonstrations divided over seven general topics. For each group, a general introduction with basic references is provided in the booklet. For each demonstration within a group a detailed background, explanation and further literature references are given. Verbal introductions on the CD have been kept very brief and serve mainly to identify each demo. Demonstrations or parts thereof can be called individually in any order, or preprogrammed in arbitrary groups, through 80 separate track or “title” numbers. One thousand copies of the CD set have been made on a first production run. They are available through the headquarters of the ASA.

A CD player with the auditory demonstration will be available at the registration desk for the entire meeting.
Principal curvatures of general wavefronts and of reflecting or refracting surfaces
Authors: Dean Cleon E. , Arnott W. Patrick , Marston P. L.
Published: October 2005

A nonparametric derivation of the exact form of the two principal curvatures κ1,2 of general wavefronts and reflecting or refracting surfaces was made for Cartesian and polar coordinates. The standard expression for the Gaussian curvature κ1κ2 was recovered in the Cartesian case. For the polar case let H = 1r WrrWr + 1r2 (WrrWθθ − Wrθ2) + 2r3 WrθWθ − 1r4 Wθ2, B = 1r Wr3 + 1r2 Wr2Wθθ + 2r3 WrWθ2 − 2r2 WrWθWrθ + 1r2 WrrWθ2 + Wrr + 1r Wr + 1r2 Wθθ, and C = 1 +Wr2 + (1/r2) Wθ2, where z = W(r,θ) is the polar equation of the surface and Wr = ∂W/∂r, Wrr = ∂2W/∂r2, and so forth; the analysis gives κ1,2 = [B ± (B2 − 4HC)1/2]/(2C3/2). The Gaussian curvature is κ1κ2 = H/C2; κ1 and κ2reduce to previous results [D. G. Burkhard and D. L. Shealy, Appl. Opt. 20, 897–909 (1981)] in the special case z = W(r). These results should be useful for generalized ray tracing and for locating caustics in catastrophe acoustics and scattering calculations. [Work supported by ONR.]
Harmonic angular perturbation of a toroidal wavefront: A simple unfolding of an axial caustic
Authors: Arnott W. Patrick , Marston Philip L.
Published: October 2005

Axial caustics can be observed in backward and forward scattering from highly symmetric surfaces such as spheres. A wavefront that is locally toroidal, propagates to produce an axial caustic. The unfolding of the axial caustic generated by a harmonic perturbation of the wavefront shape in the aximuthal direction will be considered. The perturbed wavefront shape W is given by W(s,ψ) = [(s − b)2/2α] + Λ cos (2α), in polar coordinates (s,α), where α and β specify the characteristic radii of the unperturbed torus and Λ is a measure of the perturbation. The caustic associated with W consists of four connected transverse cusps in a distorted diamond shape known as an asteroid. The caustic shape has been identified for observation points from the near zone to the far zone. Locations on W that contribute rays near a cusp point, have been identified analytically. Observations of the asteroid caustic have been made by scattering light from freely rising oblate gas bubbles in water. The bubbles are oblate since they are larger than those considered in our previous study of backscattering [W. P. Arnott and P. L. Marston, 1. Opt. Soc. Am. A (in press)]. The far zone scattering exhibits an asteroid caustic decorated by a diffraction pattern. The calculated caustic agrees favorably with observations. This calculation should also be useful in acoustic reflection and scattering problems. [Work supported by ONR.]
Acoustical phase conjugation experiments: The generation of a reversed wave through three-wave mixing in a layer of stabilized microbubbles
Authors: Kargl Steven G. , Marston P. L.
Published: October 2005

A phase-conjugate mirror is one that reverses an incident wave front so that it propagates back toward the source. Recent experiments [Kustoy et al., Sov. Phys. Acoust. 32, 500–504 (1986)] indicate that wavefront reversal can be established through the interaction of a pump wave of frequency f1 with a probe wave of frequency.f2 <f1 that diverges from a point source. Three-wave (nonlinear) mixing occurred in a layer of freely rising gas bubbles in water so as to produce a reversed wave having a frequency f3 = f1 − f2. The present research yields evidence of reversed wave generation resulting from three-wave mixing in a layer of stabilized microbubbles. One method of stabilization is to use the gas-filled micropores of a Nuclepore ® polycarbonate membrane. Our experiments were carried out in a water tank of diameter 164 cm and with frequencies f1 and f2 > 300 kHz so as to avoid spurious boundary reflections. The previously predicted [P. L. Marston, J. Acoust. Soc. Am. Suppl. 1 82, 12–13 (1987)] longitudinal and transverse focal point shifts for the reversed wave are investigated. The signal for the reversed wave at frequency f3 is enhanced through the use of a background subtraction technique. [Work supported by ONR.]
Nonlinear equations of acoustics in inhomogeneous fluids
Authors: Tjøtta Jacqueline Naze , Reiso Edel , Tjøtta Sigve
Published: October 2005

The propagation of finite amplitude sound waves produced by real sources in an inhomogeneous and thermoviscous fluid is considered. A governing nonlinear equation in the sound pressure amplitude is derived using the methods of singular perturbations. It consistently accounts for the effects of diffraction, dissipation, nonlinearity, and inhomogeneity, and represents a generalization of the parabolic equation valid for a homogeneons fluid (Khokhlov-Zabolotskaya-Kuznetsov equation) discussed in a previous work [Naze Tjøtta and Tjøtta, J. Acoust. Soc. Am. 69, 1644–1652 (1980)]. The equation also applies to the case of sound beams produced by strongly curved sources, for example, focusing and defocusing sources. The relationship to the equations of classical ray theory is discussed. [Work supported by The Norwegian Research Council for Sciences and Humanities (NAVF), the IR&D program of ARL:UT, and VISTA/STATOIL, Norway.]
Measuring sound transmission through suspended ceilings
Authors: Quirt J. D. , Halliwell R. E.
Published: October 2005

In most modern office buildings, interoffice partitions stop at the suspended ceiling. Sound insulation between such offices is generally inadequate for speech privacy. This paper presents the first stages of a project to develop practical acoustical solutions. A new laboratory has been constructed; this simulates a pair of adjacent offices with a common plenum space above the suspended ceiling. Characteristics of the test environment and pertinent standard test procedures have been studied. A series of suspended ceiling systems has been tested. The results indicate the relative importance of major physical factors controlling sound transmission through typical suspended ceilings.
Test results for floor and ceiling modifications for impact isolation
Authors: Daly Edward A. , Standlee Kerrie G.
Published: October 2005

The impact solution evaluation methods presented by the two authors in a paper given at the 112th Meeting of ASA [J. Acoust. Soc. Am. Suppl. 1 80, S56 (1986)] were used to evaluate modifications of floor/ceiling systems. Tests were conducted on the presently most-used floor structure, the wood/concrete structure. Some of the floor modifications produced significant impact sound isolation improvements. However, the floor modifications that produced the best sound attenuations provided unstable flooring conditions. All ceiling modifications produced significant impact sound isolation improvements. The type of construction required for all ceiling modifications were found to be acceptable to the construction trades. An evaluation method developed by one of the authors was used to produce a single number effectiveness rating for the floor/ceiling modifications. The results of these tests indicate ceiling modifications are more effective than floor modifications in providing impact isolation.
Laboratory calibration of standard and measurement condenser microphones
Authors: Nedzelnitsky Victor
Published: October 2005

The reciprocity technique is the most highly developed and most widely used primary calibration method at the major national standards laboratories of the world. Critical measurements of sound pressure in air are usually traceable to such calibrations, which are briefly described with reference to specific apparatus and procedures at the NBS. Secondary methods, including those based upon reciprocity, can offer improved convenience with minor to significant sacrifices in accuracy and precision. Dynamic signal analyzers incorporating digital signal processing are now sufficiently accurate and precise to offer promise for improved and/or more readily automated measurements, especially secondary free-field calibrations. Specific examples from the NBS research and measurement services are discussed.
Intensity measurement microphones and their calibration
Authors: Frederiksen Erling
Published: October 2005

New microphones especially designed for intensity measurements have made it possible to make valid intensity measurements under sound field conditions that previously would lead to invalid results. The achieved improvements are most important at low frequencies but may also be significant, for instance, at 500 Hz, depending on the sound field. The microphones of this new type are partly characterized by a stability and spread in their low-frequency phase characteristics that are a factor of 10–15 times better and partly by a sensitivity to sound pressure at the static pressure equalization vent that is several orders of magnitude lower than that of conventional microphones. These properties are all important for the improvement of intensity instruments that are based on microphones operating on the pressure principle, as the phase difference between exactly spaced points in sound fields need to be measured with great accuracy. System phase errors of the order 0.01 deg are of interest and seem to become possible by proper calibration and correction. The use of conventional microphones implies significant erors due to pressure gradients errors that are eliminated by the extremely low vent sensitivity of the new microphones. Most leading laboratories have spent great efforts on magnitude calibration of microphone sensitivity but very little on phase calibration. The microphones and the methods used for verification of microphone and probe properties are described in this paper.
Study on the linearity of laboratory standard condenser microphone
Authors: Yoshikawa S.
Published: October 2005

Because of the many features such as wide frequency range, high stability, good transient response, etc., the condenser microphone has been used as a laboratory standard microphone and a measurement microphone. One problem is that the condenser microphone has essentially nonlinear characteristics on the electrical output/acoustical input function. As sound-pressure level becomes higher, nonlinearity becomes larger. In this study, linearity of the laboratory standard condenser microphone at high sound-pressure level was investigated. In the case of the 12-in. condenser microphone MR-112, linearity was ensured up to 120 dB SPL. Over 120 dB SPL, output/input function becomes larger and distortion becomes detectable. By experiment, measured distortion factor was 2% at 151 dB SPL and 5% at 154 dB SPL. The nonlinearity at high sound-pressure level was studied analytically, and the effect of some parameters such as shape of back electrode, biasing voltage, and stray capacitance was clarified. Based on the analytical study, the possibility of lowering the nonlinearity was investigated. A method to realize the condenser microphone having low distortion and high sensitivity was suggested.
Measurement accuracy of a one-dimensional four-microphone sound intensity probe
Authors: Suzuki Hideo , Anzai Masazo , Ono Takahiko
Published: October 2005

In the sound intensity measurement, it is important that a microphone probe does not disturb the sound field. A side-by-side or a face-to-face configuration is most commonly used for the microphone probe. With these conventional configurations, however, two tubes containing individual preamplifiers are parallel and close to each other and cause a diffraction problem. A recently developed one-dimensional four-microphone probe is structurely much simpler than conventional probes. Two microphone pairs cover low- (50–800 Hz) and high- (800-10 kHz) frequency ranges and, therefore, it is possible to measure the whole frequency range without changing the microphone distance. The effective distance between pair microphones, pressure, particle velocity, and active intensity responses are measured in an anechoic chamber. Assuming a farfield condition, the measurement accuracies of pressure, particle velocity, and intensity responses are checked by comparing these with the pressure of a reference microphone. The accuracy of the reactive intensity measurement is checked by measuring the pressure square gradient by a reference microphone. The results will be shown at the meeting.
Mounting and selection of microphones for two-microphone impedance measurements
Authors: Ebbitt Gordon , Christensen Jørgen
Published: October 2005

The determination of acoustic impedance as specified in ASTM E-1050 requires the measurement of sound pressure at two positions along a standing wave tube. The choice of microphones and their mounting can have a significant influence on the measurement results. A variety of microphone types (i.e., phase-matched condensers, probe tube types, condenser microphones fitted with phase correctors, etc.) will be considered and the results from these various types will be compared. The mounting of the microphones (i.e., flush with the tube walls, near the middle of the tube, etc.) will also be considered.
Field calibration and verification of sound intensity systems
Authors: Ebbitt Gordon , Frederiksen Erling
Published: October 2005

A two-microphone sound intensity system is, in general, calibrated by using a pistonphone or a sound level calibrator to set the sensitivities of the two measurement channels. While this is all that is required for a calibration of the system, it does not provide any check of the systems performance. Such checks have traditionally been limited to the laboratory because they require special acoustic environments. A new device for calibrating and checking intensity systems will be considered here. This device consists of an acoustic source and a coupler. The coupler can be configured in two ways. In one configuration both microphones are exposed to the same sound field. This mode can be used to calibrate the channel sensitivities and to measure the residual intensity index of the system. In the other configuration the coupler will simulate free-field conditions. Either a pistonphone or a broadband source can be used as the acoustic source. The design of the device will be considered and its use illustrated.