bluuebunny/crossref_metadata_2025 huggingface dataset:
Observations of a geoclutter feature in the straits of Sicily
Authors: Weber Thomas C. , Holland Charles W. , Etiope Giuseppe
Published: January 2006

Several persistent sonar clutter features were observed in 2002 during an experiment in the Straits of Sicily (Malta Plateau) in a region that is nominally flat and thickly sedimented. High frequency sidescan and seismic reflection data from the region indicated that mud volcanoes were present and were possibly expelling clouds of methane bubbles. In 2004 the region was revisited with oceanographic equipment including a multibeam sonar, an acoustic Doppler current profiler (ADCP), CTD’s, temperature loggers, a methane sensor, and a video camera. The visual observations at one site show carbonate mounds comprised of large heterogeneous blocks that suggest the presence of mud volcanism. A multibeam sonar survey of the area shows that the carbonate mounds are of order 5–10 m in height and 10–100 m in lateral dimension, and that they form in clusters. The multibeam backscatter data show high amplitudes in the region surrounding the carbonate mounds, possibly indicating the surficial extent of the carbonate material. High levels of methane were observed in the water column above the mounds, although conclusive evidence of the presence of methane bubble clouds was not found. However, high frequency backscatter recorded on the ADCP above multiple carbonate mounds showed plume-like features that may be bubble clouds, suspended fine-grained sediment, and/or schools of fish.
Procedures which alter the intelligibility of voiced stop consonants for hearing impaired listeners
Authors: Dorman M. F. , Lindholm J. M. , Hannley M. T.
Published: January 2006

To assess the hypothesis that spread of masking from F1 is responsible for poor identification of voiced stop consonants by hearing impaired listeners, /bdg/ were synthesized with bursts and formant transitions in configurations with and without the first formant. Synthesis in the /eI/ environment allowed minimal acoustic differences to cue the contrasts. To maximize the probability of finding evidence of masking spread the stimuli were presented at levels both above and below the point of maximum intelligibility. At no point on the function was performance without the first formant better than performance with the first formant. To investigate other ways in which the intelligibility of /bd/ could be enhanced, we altered, in separate manipulations, the amplitude and duration at signal onset of the cues for place of articulation. Increasing the amplitude of the onset cues improved identification accuracy for /d/ but not for /b/ or /g/. In contrast, increasing the duration of the onset cues improved identification accuracy for /g/ but impaired accuracy for /b/ and /d/. These results and others suggest that the beneficial effects of prolonging the onset spectrum for /g/ results not from better frequency resolution due to longer processing time, but from the cue of duration per se.
Identification of words and manner features of speech by two profoundly deaf teenagers using a tactile vocoder
Authors: Brooks P. L. , Frost B. J. , Mason J. L. , Gibson D. M.
Published: January 2006

Two prelingually profoundly deaf teenage girls were taught to identify “live-voice” English words while wearing a tactile voder. To ensure the subject's deafness, hearing aids were removed and earplugs and headphones carrying white noise were worn. The reader sat out of sight from the subject so lipreading information could not be obtained. Words were introduced two or three at a time until the subject reached criterion on a 50-word list. The two subjects acquired a tactual vocabulary of 50 words with 21 and 25 h of training, respectively. Immediately after these word identification tests, experiments were carried out to test the subject's ability to identify the “manner of articulation” features of speech. Subjects could identify whether CV's contained a nasal, liquid voiced stop, unvoiced stop, or unvoiced fricative with accuracies ranging from 75% to 100% correct.
Mixed-mode acoustic glory: Model and experimental verification
Authors: Williams Kevin L. , Marston Philip L.
Published: January 2006

Scattering from large elastic or fluid spheres is enhanced along the backward axis due to a weak focusing. The effect of diffraction on this axial focusing was shown [P. L. Marston and D. S. Langley, J. Acoust. Soc. Am. 73, 1464–1475 (1983)] to remove an unphysical divergence of scattered pressure predicted by elementary ray acoustics. Our previous model of this “acoustic glory” of elastic spheres was complete only for transmitted waves within the sphere which were either all longitudinal (L) or all shear (S). These contributions were separated by measuring the backscattering of short tone bursts from a glass sphere in water having ka≈457. Measured peak-to-peak pressures were typically within 5% of predictions [P. L. Marston, K. L. Williams, and T. J. B. Hanson, J. Acoust. Soc. Am. 74, 605–618 (1983)]. In the present research we complete the theory for the case of mixed-mode glories in which there is at least one L to S or S to L mode conversion due to reflection. A novel calculation shows the focal parameters are independent of permutations of the mode sequence; however, the contributions to the scattering depend on sequence. Measured mixed-mode pressures (from the aforementioned sphere) are within 5% of the predictions. [Work supported by ONR. Marston is an Alfred P. Sloan Research Fellow.]
Temporal effects of vowel production on listeners' perception of speaker age
Authors: Oyer Herbert J. , Trudeau Michael D.
Published: January 2006

Twenty-four speakers prolonged the vowels /a e i o u/ and read the passage “My Grandfather.” The speakers were divided into four groups of six (three males, three females) based on chronological age, 41–50 years, 51–60 years, 61–70 years, and 71 years and over. Naive listeners provided perceptual judgements of each speaker's age based on the first two sentences from “My Grandfather.” The speakers' vowel productions were measured to determine time required to initiate and terminate each vowel. These ten measures plus time to articulate the four syllables /ma ɪ grændfaðɚ/ and speaker sex were regressed against speaker chronological and perceived ages. The best significant model for chronological age yielded an R-Square of only 0.428; while for the best significant model for perceived age R-Square was 0.647. Results indicate that speaker temporal control of speech exerts a strong influence on listener perception of speaker age, but has a relatively weaker relationship with speaker's chronological age. These findings suggest that temporal aspects of speech are critical in estimation of speaker age.
Stapedius reflex: On some fundamental temporal characteristics
Authors: Zwislocki Jozef J.
Published: January 2006

Effects of neural temporal summation and adaptation on the stapedius reflex are examined with the help of monotic and dichotic pairs of tone bursts. The tone intensities were so chosen that each burst presented separately evoked a constant response. The results show that the apparent summation of the effects of dichotic bursts is greater than that of monotic bursts and decays monotonically as the interburst time interval increases. With monotic bursts the apparent summation effect has a flat maximum around a time interval between the burst onsets of about 60 ms. The dichotic as well as the monotic reflex arc contains at least one nonlinearity since the summated effect of the bursts exceeds the arithmetic sum. The measured characteristics can be accounted for quantitatively by a model containing a peripheral linear adaptation stage with a recovery time constant on the order of 50 ms, a linear central temporal integrator with a time constant of about 200 ms and a threshold, all connected in series. All the numerical constants are consistent with those found in neurophysiological and psychophysical experiments. The inference that the threshold follows the integrator may prove to be of major interest.
Masking of the /da/-/ga/ distinction in cochlear impairment
Authors: Buus Søren , Miller Joanne L. , Scharf Bertram
Published: January 2006

Discrimination between two 300-ms synthesized syllables, /da/ and /ga/, presented in the middle of a 900-ms narrow-band noise, was measured in three normal and six cochlearly impaired listeners. The segmental distinction was conveyed by the frequency of the initial burst and the form of the third-formant transition, which was 50 ms in duration: For /ga/ the transition started around 1.9 kHz and rose to its steady-state value of 2.5 kHz and for /da/ it started around 3.0 kHz and fell to 2.5 kHz. The narrow-band noises were 300 Hz wide and were centered between 1 and 3.5 kHz. For the normal listeners, the speech was set to 70 dB SPL and the noise to 80 dB SPL. For the impaired listeners, the levels were chosen to yield better than 95% discrimination in the easiest condition, but poor discrimination in the most difficult condition. Both groups of listeners showed the poorest performance when the noise was centered at 2.5 kHz. As the noise moved away from this critical region performance improved for both groups, but more rapidly for the normal listeners. These results indicate that the impaired listeners' reduced frequency selectivity may contribute to their difficulty of discriminating speech in noise. [Supported by NSF and NIH.]
Document printer source noise control—A case history
Authors: Lilley Daniel T.
Published: January 2006

Noise reduction of at least 17-dB A-weighted sound level was obtained on a multi-function, dot matrix document printer by incorporation of damping, vibration isolation, and modification of the acoustical paths within the machine. This result was achieved without the use of sophisticated measurement and analysis techniques, but by consideration of basic principles of structural dynamics, application of appropriate materials in conjunction with minor design modifications, and attention to detail. Method of measurement was per ANSI S1.29-1979, “Method for the Measurement and Designation of Noise Emitted by Computer and Business Equipment.” Emphasis is given to the importance of using highly damped isolation materials which permit large impedance mismatches at structural attachment sites while maintaining integrity of static connections. A discussion of noise sources and treatment approach for each is provided.
Electroglottographic and acoustic waveforms of voice onset in stutterers
Authors: Borden Gloria J. , Baer Thomas
Published: January 2006

The first few glottal pulses in the stuttered and fluent tokens of voiced utterances by stutterers were analyzed and compared with the initial voice pulses in the same utterances by nonstutterers. Signals analyzed were the impedance changes across the glottis during voice initiation via electroglottography (EGG) and the acoustic waveforms. In single-cycle analysis of vocal fold vibration, normal speakers evidenced abrupt decrease in impedance with increasing vocal fold contact, and a more gradual increase in impedance as the folds opened. Stutterers, when perceived as fluent, followed the same pattern as normals. Extremely aberrant EGG patterns often accompanied stuttering episodes, with idiocyncratic strategies revealed for breaking the stuttering block. The envelope for the first EGG cycles differed from normal for some stutterers. Some of the severe stutterers showed patterns of gradual buildup of EGG amplitude after a block, whereas the mild stutterers and normal controls showed a more abrupt envelope. When fluent, all speakers had EGGs with abrupt envelopes and a more stable open phase for each cycle. EGG activities will be related to their acoustic correlates. [Work supported by NIH.]
High-resolution beamforming with oversampled arrays
Authors: Byrne Charles L. , Fitzgerald Raymond M.
Published: January 2006

Plane-wave beamforming is interpretable as estimation of the Fourier wave vector spectrum from samples of the spatially dependent acoustic field. Because spatially localized sources contribute discrete peaks to the wave vector spectrum, the use of high-resolution nonlinear techniques (such as Burg's maximum entropy method—MEM) to resolve closely spaced sources has received much attention [Johnson, Proc. IEEE 70(9), 1018–1028 (1982)]. Such techniques are expected to be most useful for short arrays, whose dimensions do not exceed a few wavelengths, for which case conventional beamforming offers poor resolution. Such short arrays typically contain a number of hydrophones and thus oversample the spatial field; the spacing between adjacent hydrophones is less than one half a wavelength. When MEM is applied to oversampled arrays, spurious features are often produced due to the concentration of the spectrum within the visible region of wave vectors. These spurious features can be eliminated through the use of a more general high resolution procedure, the weighted reciprocal spectrum approximation method [Byrne and Fitzgerald, IEEE Trans. Acoust. Speech Signal Process. ASSP-31, 276–279 (1983)].
A comparison of spatial spectra from vertical line arrays of different aperture using maximum entropy method (MEM) beamforming
Authors: Lawrence Mimi Zebrick , Ramsdale Dan J.
Published: January 2006

This study was undertaken to provide a preliminary answer to the question of how aperture size affects beamwidth and resolution for vertical line array data which is processed using the (MEM) beamformer. This was done empirically by examining the (MEM) beamformed output of a nine-element vertical line array and reducing the aperture by deleting hydrophones. Narrow-band acoustic data were provided by a cw source (54 Hz) which was towed at a nominal depth of 91 m radially away from the array. The (MEM) beamformer was implemented using the Burg algorithm in complex form with averaging performed over the filter coefficients. The spatial spectrum was characterized in general by two dominant arrivals which varied in angular separation and level relative to the background noise. For arrivals with high signal-to-noise ratios and large angular separation, preliminary results showed little degradation in beamwidth as the aperture was decreased by a factor of two. As the angular separation between arrivals decreased or when the signal-to-noise ratio decreased, beamwidth and resolution were degraded when the aperture was reduced.
New applications of nonlinear acoustics at microwave frequencies
Authors: Rugar Daniel
Published: January 2006

Three important new applications have been found for nonlinear acoustics in the microwave frequency regime. For the first example, the gigahertz frequency acoustic microscope is seen to significantly increase in resolution when operated at high transmitted power levels. This effect is believed to be due to energy flow from the second and higher harmonics back to the fundamental frequency after passage through the focal region of the beam. Using this effect, resolution better than 0.2 μm has been demonstrated in water at 4.3 GHz [B. Hadimioglu (to be published)]. The second application of nonlinear acoustics uses the focused beam of a liquid helium acoustic microscope to detect incoherent phonons (10 GHz) generated by a heater [J. S. Foster, D. Rugar, and C. F. Quate (to be published)]. This technique, which relies on scattering of sound by sound, has resulted in a new form of acoustic imaging. The third new application has provided the first accurate measurements of low-energy phonon dispersion in liquid helium at temperatures less than 0.1 K. For plane waves in a dispersive, lossless fluid, power in the second harmonic will periodically rise and fall as a function of propagation distance. By measuring the periodicity, dispersion can be accurately determined. This has recently been accomplished at a fundamental frequency of 2 GHz. [Work supported by ONR.]
Metal fatigue testing at ultrasonic frequencies, history and uses
Authors: Mason Warren P.
Published: January 2006

Ultrasonic frequencies in the range of 20 kHz are now being used to test fatigue in materials on account of the rapidity with which the tests can be made. It appears that the original work was done by the speaker as described in the paper. At a recent international conference “Fatigue and Corrosion Fatigue up to Ultrasonic Frequencies,” which was dedicated to me on account of my original work in the field, it is stated that as a technique ultrasonic fatigue is almost as sophisticated as push-pull fatigue testing. It has been shown that the results obtained at high frequencies are closely similar to results obtained at low frequencies, cycle per cycle. Constant total strain and constant displacement tests are possible. Positive mean load, elevated temperatures, and corrosion ultrasonic fatigue tests are now performed routinely. The feasibility and advantages of performing crack propagation tests at ultrasonic frequencies has been demonstrated. Hence ultrasonic fatigue testing is a very widely used technique.
Surface statistics and the spatial coherence of surface-scattered energy
Authors: Kinney W. A. , Clay C. S.
Published: January 2006

The facet-ensemble method [W. A. Kinney et al., J. Acoust. Soc. Am. 73, 183–194 (1983)] was used to determine how the statistics of a rough surface relate to the spatial coherence of energy scattered from it. Two ideal water surfaces were studied. One had a sawtooth cross section derived from the sum of the first four terms of a Fourier sine series representation. The cross section of the other surface was derived from the sum of the same terms but with each having random phase. This resulted in two surfaces each with the same rms roughness (0.2 cm ), spatial wavelength (16 cm), and spatial wavenumber components, but different cross-sectional shapes. With each surface, the acoustic field in water was computed at two receiver locations for both backscatter and forward scatter for a grazing angle of 35 (with respect to the surface mean), 100 kHz, and a total travel distance of 421 cm. For backscatter, the pressure amplitudes and coherence values differ considerably between the two surfaces, and the differences increase with increasing receiver separation. For forward scatter the differences are much less. [Work supported by NORDA.]
High-frequency scattering from the sea surface
Authors: McDaniel Suzanne T.
Published: January 2006

Composite roughness theory provides one means of treating high-frequency scattering from the sea surface. In this method, the surface is artificially partitioned into two wavenumber regimes leading to results that are dependent on precisely how the surface is partitioned. In this paper, diffractive corrections to the geometrical optics solution of the Kirchhoff integral are derived. The correction terms, in this case, arise quite naturally from an asymptotic expansion of the scattering integral and are, hence, not dependent on the choice of an arbitrary parameter. Numerical results are obtained for a two-dimensional surface and compared with the results of composite roughness theory. [Research supported by NORDA.]
Comparisons between sinusoidal boundary scattering theories and measured data
Authors: McCammon Diana F.
Published: January 2006

An exact solution to scattering from a sinusoidal boundary has been presented by Holford [J. Acoust. Soc. Am. 70, 1116–1128 (1981)]. This solution, and the approximate solutions of Kirchhoff, Eckart, Brekhovskikh, and Rayleigh are contrasted with the measured data given by LaCasce and Tamarkin [J. Appl. Phys. 27 [2] (1956)] for three different surface roughnesses. Very good agreement is obtained by the Holford theory, especially in the location of reflection amplitude peaks and zeros.
Application of the composite roughness model to bottom backscattering
Authors: Jackson Darrell R. , Winebrenner Dale P. , Ishimaru A.
Published: January 2006

Ocean bottom acoustic backscattering is modeled as an incoherent sum of intensifies due to interface roughness and volume inhomogeneity. Rough interface scattering is treated by a version of the composite roughness model similar to that applied to the sea surface [S. T. McDaniel and A.D. Gorman, J. Geophys, Res. 87, 4127–4136 (1982)]. The volume scattering portion of the model employs the first-order multiple-scattering approximation including refraction and large-scale interface slopes, but neglects layering [A. N. Ivakin and Yu. P. Lysanov, Sov. Phys. Acoust. 27, 212–215 (1981)]. The model is compared with data presented at the Cincinnati meeting [J. J. Crisp, P. A. G. Thomas, A. M. Baird, and D. R. Jackson, J. Acoust. Soc. Am. Suppl. 1 73, S97 (1983)] using power-law fits to measured bottom relief spectra. The relatively slow fall-off of these spectra causes the composite roughness model to be sensitive to the large-scale cutoff wavenumber. [Work supported by ONR.]
Sonar estimates of sea floor microroughness
Authors: Stanton T. K.
Published: January 2006

This paper presents an analysis of the effects of microroughness of the ocean bottom on a sonar signal. The results best apply to features where the roughness amplitude is less than one-quarter of an acoustic wavelength such as with ripples, beds of rocks, and nodules. The shape of the probability density function (PDF) of the echo envelope is examined in terms of the rms roughness and a new parameter, the correlation area of the bottom. The area is equal to the product of the x and y correlation distances along the floor. The PDF is shown to be extremely sensitive to small changes in the roughness. Furthermore by determining the rms roughness from standard coherent reflection measurements, the correlation area may be extracted directly from the PDF. Thus both vertical as well as lateral information is obtainable from sonar data. The technique can be used to discriminate between different types of bottoms that may have the same roughness but different correlation areas, for example a floor with ripples versus one with rocks or nodules. The analysis combines two approaches: (1) A general statistical model employing the Rice PDF (S.O. Rice, in Selected Papers on Noise and Stochastic Processes, edited by N. Wax (Dover, New York, 1954), pp. 133–294] and (2) a theory originated by Eckart [C. Eckart, J. Acoust. Soc. Am. 25, 566–570 (1953)]. The analysis is applied to sonar data collected from the continental shelf near Cape Hatteras, North Carolina. The results are consistent with the known characteristics of the area.
An overview of shock and vibration measurements
Authors: Volin Rudolph H.
Published: January 2006

Early instruments limited our ability to make shock and vibration measurements; however, the development of new instruments and measuring techniques has advanced the state of the art in shock and vibration measurement. The most significant developments were the piezoelectric accelerometer and the Fast Fourier Transform algorithm. Both of these developments, coupled with advances in the digital computer technology, significantly influenced the current state of the shock and vibration measurement technology. Piezoelectric accelerometers enable us to measure complex motions of lighter structures to higher frequencies. The Fast Fourier Transform algorithm initially made it possible to analyze shock and vibration test data on main frame digital computers rapidly and economically; but, combined with advances in the digital computer technology, it has broadened our ability to conduct many types of laboratory shock and vibration tests, make measurements, and analyze test data. Advances in the digital computer technology will continue to influence the progress in shock and vibration measurement. For example, the microcomputer has already proven to be effective in certain data analysis applications. Future increases in their computing power will allow them to be used in a wider range of measurement and data analyses applications.
Underwater modal analysis of heat exchanger tubing using pressure transducers
Authors: Powell C. D. , Dietz C. P. , Middelburg N. J.
Published: January 2006

In order to perform an accurate flow-induced vibration analysis of heat exchanger tubing, natural frequencies, mode shapes, and damping values must be known. This study concerns experimental determination of the above parameters for titanium tubing in an underwater environment. Being a lightweight material with high stiffness, it was anticipated that high natural frequencies with small damping values would be encountered which could be erroneously altered by the mounting and cabling of traditional instrumentation such as accelerometers. A 1-g microaccelerometer was used to perform a preliminary modal survey to determine system natural frequencies, mode shapes, and separation of tube and fixture modes. It was found that mounted instrumentation did indeed cause mass loading problems resulting in erroneous natural frequencies and damping values. Noncontacting pressure transducers were used to confirm the identical tube mode shapes and to then determine true tube natural frequencies and damping values using a variety of excitation techniques. The study concludes by evaluating the most applicable techniques to be used in order to obtain valid data.
Transient waves in an elastic plate
Authors: Proctor T. M. , Breckenridge F. R. , Pao Y-H.
Published: January 2006

Recently Pao et al. [Y-H. Pao, R. R. Gajewski, and A. N. Cevanoglu, J. Acoust. Sac. Am. 65, 96–105 (1979)] have obtained theoretical solutions for wave propagation in an elastic plate by using generalized ray theory. These solutions give the dynamic displacement at locations on the surface of a plate due to a point force step function input. Using the NBS conical transducer, a broadband displacement transducer, experimental verification of the motion has been obtained.
On the use of hidden Markov models for speaker-independent recognition of isolated words from a medium size vocabulary
Authors: Rabiner L. R. , Levinson S. E. , Sondhi M. M.
Published: January 2006

Recent work at Bell Laboratories has shown how the theories of LPC Vector Quantization (VQ) and hidden Markov modeling (HMM) can be applied to the recognition of isolated word vocabularies. Our first experiments with HMM based recognizers were restricted to a vocabulary of the ten digits. For this simple vocabulary we found that a high performance recognizer (word accuracy on the order of 97%) could be implemented, and that the performance was, for the most part, insensitive to parameters of both the Markov model and the vector quantizer. In this talk we extend our investigations to the recognition of isolated words from a medium size vocabulary, (129 words), as used in the Bell Laboratories airline reservation and information system. For this moderately complex vocabulary we have found that recognition accuracy is indeed a function of the HMM parameter (i.e., the number of states and the number of symbols in the vector quantizer). We have also found that a vector quantizer which uses energy information gives better performance than a conventional LPC shape vector quantizer of the same size (i.e., number of codebook entries).
A finite state parser for use in speech recognition
Authors: Church Kenneth W.
Published: January 2006

The application of finite-state parsing techniques is proposed for the purpose of assigning syllable structure to a phonetic transcription, subject to allophonic and phonotactic constraints. In this way, a speech recognition device can exploit contextually VARIANT cues in order to distinguish minimal pairs such as “a Tease/aT ease,” “nighT Rate/niTRate,” and “greaT Wine/gray TWine.” Our parsing suggestion provides an elegant implementation framework for exploiting the important interaction between segmental variation and syllable/word context. Other examples will be discussed, including Klatt's “Did you hit it to Tom?” Finally, the framework will be extended to account for “feature spreading” (agreement and coarticulation) in a natural way, by reformulating parsing in terms of four matrix operations: addition, multiplication, transitive closure, and element-wise intersection.
A comparison of dynamic time warp algorithms for connected word recognition
Authors: Glinski Stephen C.
Published: January 2006

The application of dynamic time warping to the problem of connected word recognition has recently received much attention. Several successful approaches have appeared in the literature including the level building algorithm [C. Myers and L. Rabiner, IEEE Trans. Acoust. Speech Signal Process. ASSP-29, 284–297 (1981)] and the single pass algorithm [J. Bridle, M. Brown, and R. Chamberlain, Proc. 1982 IEEE ICASSP]. A single mathematical description of dynamic time warping is presented that unifies these and other approaches, and highlights their similarities and differences. By testing the algorithms with a database of connected digits, it is found that in general, the use of additional levels of computation results in a relatively small decrease in recognition error.
Vowel-space normalization procedures and phone-preserving transformations of synthetic vowels
Authors: Nearey T. M.
Published: January 2006

A variety of speaker-normalization procedures have been proposed for the recognition of vowels of different speakers and/or the cross-language comparison of vowel systems. Many such procedures are special cases of affine transformations of coordinates in a two-(or more) formant space. Results of perceptual experiments are presented indicating that only a limited subset of affine transformations preserve the phonetic identity of a set of synthetic vowels. From a perceptual point of view, many vowel normalization procedures appear to be too powerful in the sense that they can “over-normalize”" vowels which are phonetically distinct. In particular, rotations of a two-formant space and/or independent multiplicative scaling of the F1 and F2 axes lead to marked change in phonetic quality. On the other hand, results indicate that uniform multiplicative sealing of F1 and F2 by a single constant is “almost phone-preserving” in the range of male/female vowel differences. For larger (e.g., male/child) differences, a specific nonuniform scaling similar to that proposed by Fant seems to be required to maintain “perceptual constancy.”
Changing conceptions of pitch structure and timbre: A modest proposal
Authors: Balzano Gerald J.
Published: January 2006

Two areas of musical exploration of interest to composers for which the computer is unusually well-suited are microtonal systems and “new” timbres. Notions about how to achieve these musical extensions are doubly theory-dependent in that they depend on both our theory of perception and our theory of the stimulus. Most extant attempts to do microtonal music have used small-integer ratios as the fundamental description of the musical elements involved and the basic entities to which human perceivers are presumed sensitive. For timbre, the universe of possibilities has traditionally been characterized in terms of spectral variables (possibly time-varying), and more recently in terms of projections on axes in a “subjective” multidimensional space. In the present paper, alternatives to prevailing conceptions of both pitch and timbre are considered. A symmetry-oriented group-theoretic approach to pitch structure that eschews ratios will be described; the resulting view of pitch systems leads to some novel ideas about how to do microtonal music on a computer. For timbre, an approach that focuses on the dynamics of the sound-producing activity rather than on the resulting spectra will be described. This leads naturally to an interesting alternative to currently popular methods for generating timbres and “timbre transformations” on a computer.
Method for separation of acoustic levitation and rotation modes leading to arbitrary axes of rotation
Authors: Allen J. L. , Barmatz M.
Published: January 2006

A method for separating acoustic levitation and rotation of small objects has been developed and tested experimentally. Sample rotation in an acoustical resonance chamber requires a phase difference between two equal frequency (degenerate) modes. Therefore, this method uses one set of nondegenerate modes for levitation and higher order sets of degenerate modes for rotation. This separation of modes may be accomplished by appropriate choices of chamber dimension ratios producing up to three sets of degenerate orthogonal higher order modes. By appropriate excitation of these sets of modes, arbitrary axes of rotation may be achieved. This technique for one rotation axis was verified experimentally using a rectangular chamber. Spherical samples were stably levitated without rotation using the three orthogonal fundamental modes, and rotated about the z axis by generating a phase difference between the degenerate third and fifth harmonics of the x and y directions, respectively. A video tape of these levitation experiments will be presented. [Work supported by NASA.]
Effect of the player's vocal tract on the tone quality of reed woodwind instruments
Authors: Backus John
Published: January 2006

It has been argued by woodwind instrument players that the configuration of the player's vocal tract has an effect on the tone quality of the instrument when played. This matter is being investigated. The following arrangement has been set up: A “clarinet” is simulated by mounting a clarinet mouthpiece and reed on a piece of brass tubing of such length as to sound the note D3 when blown, using a foam neoprene “lip” with the reed. A microphone is attached to the mouthpiece to monitor its internal pressure. The open end of the brass tube is sealed onto the end of a 4-in.-diam tube 10 ft long filled with sound absorbing material. By attaching a vacuum cleaner to the other end of the 4-in. tube, the “clarinet” can be “blown,” as it were, in reverse, with the reed end out in the open. Various systems can now be attached to the mouthpiece and their effect on the internal waveform of the mouthpiece determined. Results so far indicate that the effect of the player's vocal tract on the instrument's tone quality should be negligible. [Work supported by the NSF.]
Recorder multiphonics
Authors: Ayers R. Dean , Eliason Lowell J. , Forel David
Published: January 2006

A systematic study has been undertaken to identify and analyze all possible multiphonics on a popular brand of plastic recorder (Aulos, alto voice). The absence of a bell at the end of the recorder combines with the open-hole cutoff effect to guarantee strong, nonharmonic bore resonances, and many of the forked fingerings produce stable multiphonics when blown somewhat hard. The more stable and aurally interesting multiphonics have two strong spectral components, one supported by the fundamental bore resonance and the other by the fourth or fifth resonance. Heterodyne descendants of these components are also found, but rarely with comparable strength. The less interesting multiphonics have just one strong component, often at the third resonance, with noticeably weaker components beating against it. Some fingerings will produce more than one multiphonic, depending on the breath pressure and/or attack used. One factor that may be significant for stability is the fact that the edge tone itself goes multiphonic at the blowing pressures used, without any feedback from the bore.
A reference speech recognition algorithm: An approach to characterizing speech data bases
Authors: Hieronymus James L. , Pallett David S.
Published: January 2006

One way to characterize the relative recognition difficulty of speech data bases is to use a reference speech recognition algorithm. This approach was first suggested by R. K. Moore at the Royal Signals and Radar Establishment. A preliminary algorithm for this purpose has been published [G. F. Chollet and C. Gagnoulet, Proc. 1982 IEEE ICASSP, 2026–2029 (1982)]. Currently, an isolated utterance reference algorithm under development in our laboratory includes automatic endpoint detection and a training technique which averages several tokens of each utterance. Mel scale cepstral coefficients are used in conjunction with dynamic time alignment in both training and recognition. Recognition accuracy and performance statistics are presented along with distance measures for best and second-best match. Histograms of the ratio of second-best to first-best scores are plotted as a measure of data base confusability. If the ratio histogram has considerable population near unity, there is significant danger of confusion. Other descriptors characterizing properties of the speech data base are presented. Preliminary results of research on confusability measures are reported.
One-pass, speaker-independent, isolated digit recognition system
Authors: Vysotsky George
Published: January 2006

The system described in this paper is an outgrowth of one presented earlier [J. Acoust. Soc. Am. Suppl. 1 72, S30 (1982)]. It is a combination of deterministic and probabilistic approaches to isolated word recognition. The most distinguishing features of the new system are a real-time, one-pass algorithm and the use of telephone-quality speech rather than broadband microphone speech. This new version is targeted to a single chip implementation with the requirement that the system uses no more than 256 bytes of RAM and 8000 bytes of ROM. The system makes decisions based on a single 20 ms frame often features (with no long term storage of previous frames). Preliminary segmentation and matching are combined to significantly reduce both the number of confusable words at each recognition point as well as the memory requirement for references obtained during the multispeaker training. The system was evaluated on a ten digit vocabulary (1000 tokens, 21 talkers) and showed an accuracy of 94.4% with a 3.1% error rate, and a 2.5% rejection rate. The effects of using a one-pass algorithm on total performance are also discussed.
The SIGPRO signal processing software development system
Authors: Wolitzky Jan I.
Published: January 2006

The SIGPRO system is designed to assist in the development of digital signal processing software. By freeing the user from such “bookkeeping” tasks as file management, memory allocation, control flow decisions, variable name and default parameter assignment, and error- and type-checking and correction, the SIGPRO system allows a signal processing software system to be specified solely in terms of standard “building block” modules and the connections between them. A library of approximately 100 such modules, for performing such operations as correlation, filtering and filter design, Fourier transforms, LPC synthesis, etc., is included, along with a mechanism for easily adding new modules to the system. The system is written entirely in the c programming language, and generates as its output a c program that can be compiled to produce correct implementation of the specified system, either as an executable program or as a C-callable subroutine (that may, in turn, be integrated into the module library). The program runs under virtual-memory versions of the Unix(TM) operating system. (Unix is a trademark of Bell Telephone Laboratories, Inc.)
Storage and retrieval of quality digitized long speech phrases using a microcomputer
Authors: Todd Henry S.
Published: January 2006

Two microcomputer digital speech storage and retrieval systems were designed and evaluated in a computer-assisted instruction environment. An 8-bit Z80 microprocessor was used for logic control with storage of programs, text files and digitized speech on hard disk. Speech sampling rates of 10 and 12.5 kHz were utilized for digital 8-bit (256 amplitude levels) speech storage and reconstruction. Speech phrase lengths varied from 2 to 20 s.
Fluttering target detection in horseshoe bats
Authors: Schnitzler Hans-Ulrich
Published: January 2006

The rhythmical modulations in insect echoes caused by the moving wings are behaviorally relevant information for horseshoe bats. By evaluating this information the bats are able to detect insect echoes even in strong background clutter and probably to classify different species. Transmitter and receiver of the horseshoe bats' echolocation are especially adapted for the processing of this behaviorally relevant information. An adaptation of the transmitter is the Doppler shift compensation which keeps the carrier frequencies of the insect echoes within an “expectation window” and uncouples the insect echoes from the bats' flight movement. Adaptations of the receiver are the specialized structure of the cochlea with the highly expanded frequency representation in the range of the insect echo frequencies which leads to a strong overrepresentation of sharply tuned neurons with special response characteristics throughout the whole auditory pathway. Thereby the bats establish an “analysis window” in the receiver which corresponds to the “expectation window” caused by the transmitter characteristic.
Echolocation behavior in a “flycatcher” bat. Hipposideros diadema
Authors: Brown Patricia E. , Berry Robert D.
Published: January 2006

Echolocation behavior and sonar pulse design in free-ranging Diadem horseshoe bats (Hipposideros diadema) were studied during the NSF-sponsored expedition to Chillagoe, Australia. To follow individual bats under field conditions, crystal-controlled radio transmitters were attached to the bats. This produced valuable data on home range, activity patterns, and foraging behavior as well as permitting the recording of echolocation signals of known bats under field conditions. In its foraging area, a bat would hang from a limb, acoustically scanning for a passing beetle. Upon detection, the bat briefly tracked its prey before darting out to capture it and returning to the same roost. Hipposideros diadema employs a CF/FM signal with a resting frequency of 55 kHz. They are capable of Doppler-shift compensation, altering the CF component in response to movements of nearby objects. While Rhinolophus ferrumequinum will lower its resting frequency to compensate for its own flight speed, it does not react to negative Doppler shifts caused by receding objects. Hipposideros is stationary for prey detection, and compensates for both negative and positive Doppler-shifted echoes generated by approaching and receding insects by lowering and raising the CF component of its sonar pulses.
Coherence analysis applied to structural acoustic problems
Authors: Bendat J. S.
Published: January 2006

This presentation discusses signal processing techniques and engineering requirements for solving structural acoustic multiple input/output problems using ordinary, partial, and multiple coherence functions. Practical matters are included on formulation of models, selection of input records, time delay bias errors, measurement interference effects, and statistical sampling errors. These topics are covered in the book by J. S. Bendat and A. G. Piersol [Engineering Applications of Correlation and Spectral Analysis (Wiley-Interscience, New York, 1980)]. Recent two-input/one-output models employing these methods are: (1) space shuttle tile problems where it is required to determine the response of tiles due to pressures from acoustic noise inputs over the tiles, versus that from correlated panel vibration inputs under the tiles (performed by A. G. Piersol); (2) a noise source identification problem where it is required to determine the relative contributions of a structure-borne noise input generated in an engine, versus a correlated cooling fan noise input from a fan mounted on the engine, to a farfield sound output (performed by W. G. Halvorsen).
Signal processing techniques used in underwater acoustics
Authors: Anderson V. C.
Published: January 2006

The two underlying goals of most signal processing applications in underwater acoustics are detection and estimation of a signal in the presence of uncertainties. The use of standard techniques such as matched filtering, prewhitening, and adaptive parameter estimation brings into play new dimensions of technology in the face of the wide variability and uncertainty in parameters of the signal and the background. This paper will present an overview of signal processing applications in the field.
The perception of individual “harmonics” of a ripple-noise spectrum
Authors: Burns E. M. , Corban M. , Feth L. L. , Kirby V.
Published: January 2006

A series of experiments were performed in order to compare the ability of listeners to obtain pitch information from a single “harmonic” (ripple) of a ripple-noise spectrum with their ability to obtain similar information from an equivalently filtered white noise spectrum. The experiments included: (1) a study to determine the ability of listeners to “hear out” individual ripples from a ripple noise using Plomp's paradigm [Plomp, J. Acoust. Soc. Am. 36, 1628–1636 (1964)], (2) the determination of “frequency” JNDs for individual ripples and for narrow-band noises, (3) the determination of “fundamental frequency” JNDs (by randomizing ripples within trials), and (4) musical-interval-identification studies. The results are discussed in relation to temporally based visa vis spectrally based models of ripple noise pitch perception. [Supported by NINCDS.]
Listeners' identification of human-imitated animal sounds
Authors: Lass Norman J. , Eastham Sandra K. , Wright Tammie L. , Hinzman Audrey R. , Mills Karen J. , Hefferin Amy L.
Published: January 2006

The purpose of this investigation was to determine if listeners can accurately identify human-imitated animal sounds. A total of 20 speakers, ten females and ten males, recorded their imitations of cows, cats, dogs, pigs, and sheep. A master taps containing the randomly arranged recordings was prepared and presented to 30 judges who were asked to identify the imitated animals and to rate the confidence of their judgments on a seven-point rating scale. Results indicate that: (1) listeners' accuracy was relatively high for all five human-imitated animals; (2) imitations of dogs were most accurately identified, followed by sheep, cows, cats, and pigs; (3) for all five animals investigated, listeners' accuracy was greater for female-imitated than male-imitated sounds; (4) listeners' confidence ratings were highest for sheep followed by cows, dogs, cats, and pigs; and (5) listeners' confidence ratings were higher for female-imitated than male-imitated animal sounds (with the exception of dog sounds). Implications of these findings and suggestions for future research are discussed.
Crib-o-gram (COG) and ABR effect of variables on test results
Authors: Wright Laura B. , Rybak Leonard P.
Published: January 2006

This study compared seven categories of apparently relevant information concerning the Crib-o-gram (COG) as screening test and evaluated the following variables: birth weight, gestational age at birth, Apgar scores at 1 and 5 min, apnea, ototoxic drugs, number of days on assisted ventilation, and gentamicin blood levels. Low birth weight, early gestational age, and apnea were significantly correlated with COG results. Thirteen of the 55 newborns tested failed the COG at least twice before having the ABR test given. Twelve infants had normal ABR and one had same degree of hearing loss. Consequently the rate of false positive finding is very high. In spite of the high risk factors of the newborns, gentamicin, Apgar scores, or number of days on a ventilator were not implicated as a cause of hearing loss. [Work supported by Deafness Research Foundation.]
Auditory brainstem response signal transduction through a Telephonics TDH-39 earphone
Authors: Olsen C. Charles
Published: January 2006

Waveforms and power spectra of acoustic signals used to elicit the auditory brainstem response were analyzed with respect to electrical and acoustical differences. Signals were of alternating polarity and included a dc pulse, tone bursts of 0.5 and 4.0 kHz, a 4.0-kHz tone pip, and a 2.0-kHz logon. The earphone faithfully transduced the temporal dimension of the 5.0-ms bursts and pips while the logon (450 μs) and the click (100 μs) durations were on the order of ten times greater than the electrical input. Earphone distortion included the loss of well-defined energy peaks and a spread of energy to higher frequencies. Signal polarity was not maintained through the earphone.