bluuebunny/crossref_metadata_2025 huggingface dataset:
Homophasic and antiphasic masking as a function of signal bandwidth and frequency region
Authors: Preece John P. , Eddins David A. , Wilson Richard H.
Published: October 2005

Thresholds for homophasic and antiphasic multicomponent tonal complexes were measured as a function of bandwidth in broadband noise to replicate and to extend the classic masking experiment [G. Gässler, Acoustica 4, 408–414 (1954)]. Gässler reported that masked thresholds for tonal complexes were nearly constant for bandwidths less than a nominal “critical bandwidth,” beyond which thresholds increased 3 dB/oct. Three experiments were conducted in broadband noise employing (a) homophasic signals similar to Gässler's (1 to 20 sinusoids spaced 20 Hz apart extending downward from 1100 Hz); (b) homophasic and antiphasic signals composed of 1 to 23 sinusoids centered at 500, 1000, and 2000 Hz; and (c) homophasic and antiphasic signals centered at 500 Hz with bandwidths to 800 Hz (71 sinusoids). The slope of the threshold/signal bandwidth function beyond 100 Hz was < dB/oct at each center frequency, ranging from 0.3 dB/oct (2000 Hz) to 1.9 dB/oct (1000 Hz). The data indicate that the integration of signal energy continued beyond the so-called “critical band” estimates reported by Gässler. There were no substantial differences between the slopes of the homophasic and antiphasic conditions at 500, 1000, and 2000 Hz. [Work supported by VA Medical Research Service.]
Psychometric functions for unattended tones
Authors: Dai Huanping , Buus Soren , Scharf Bertram
Published: October 2005

Listeners detect attended tones better than unattended tones. To determine how much better, percentage correct detection was measured as a function of signal level for tones at 24 unattended frequencies, all more than a critical band from the attended or primary frequency of 1000 Hz. The primary was presented as a cue at the beginning of every trial (2I, 2AFC) and also as the signal on 67% of the trials. On the rest of the trials, the signal could be any one of 24 different probe frequencies. All signals were presented against a wide-band noise at 62 dB. Nearly 6000 trials by three experienced subjects yielded a mean psychometric function for the probes that was parallel to that for the primary, but shifted to the right, showing poorer detection of the probes. Closer analysis revealed that the 14 probes at frequencies above 1000 Hz were “attenuated” 2 to 3 dB, and the 10 probes below were attenuated 4 to 6 dB. These and earlier results suggest that probe frequencies may be effectively attenuated 6 dB or more. [Work supported by NIH.]
Evaluation of a physiologically based auditory model using psychophysical data
Authors: Jeng Patricia S. , Hall Joseph L.
Published: October 2005

A physiologically based auditory model [O. Ghitza, J. Phon. 16(1), 109–123 (1988)] was tested against human psychophysical data. The model consists of a set of cochlear filters inferred from frequency selectivity of primary auditory fibers in cat, rate-level processing, and synchrony extraction. Isoloudness contours for pure tones are calculated from the model and are compared to those measured in human by Fletcher and Munson (1933) and by Robinson and Dadson (1956). The model is also used to simulate Scharf's [J. Acoust. Am. 31, 365–370 (1959)] critical-band measurements using complex sounds of four components. In addition, both calculated and measured critical-band results are plotted as a function of cochlear place using formulas for human and cat cochlear maps in Greenwood's recent paper (submitted to J. Acoust. Soc. Am.). The comparison of calculated and measured data suggests possible improvements of the model. Laboratories, Acoustics Research Department.
Effects of auditory segregation of a harmonic component on discrimination of diotic intensity and of interaural time difference
Authors: Woods W. S. , Colburn H. S.
Published: October 2005

Researchers have demonstrated that the just-noticeable difference (jnd) for interaural time difference (ITD) of a tonal target increases in the presence of tonal maskers, even with target/masker separations greater than a critical band [Buell and Hafter, J. Acoust. Soc. Am. Suppl. 1 83, S122 (1988); Stellmack et al., J. Acoust. Soc. Am. Suppl. 1 85, S83 (1989)]. In most cases, subjects report the perception of one “sound object,” whose subjective position is affected by the ITD at each frequency of the stimulus. It has also been noted that asynchronizing the onset times of components of a stimulus yields more than one “sound object.” This experiment investigates whether this parsing can lead to unmasking of a component, and whether a general model of parsing can be used to predict observed effects. A 41, 2AFC adaptive procedure was used to measure diotic intensity and ITD discrimination jnd's for a 100-ms, 600-Hz target, both in isolation and when masked by a two-tone masker comprising 400- and 800-Hz tones, under varying conditions of relative onset time of masker and target. Although generally lower jnd's were obtained for the asynchronous case, the fact that the difference was small relative to the variability in the adaptively measured jnd's prompted a fixed-increment measurement of sensitivity under the same conditions. [Work supported by NINCDS.]
Converging evidence on the nature of the segmental representation underlying spoken word recognition
Authors: Gagnon Deborah A. , Sawusch James R.
Published: October 2005

Previous work has shown that the segmental representation used in recognizing spoken words does not correspond to the traditional abstract phoneme [Gagnon and Sawusch (1989)]. However, the evidence did not unequivocally distinguish between two alternative proposals—allophones and position-specific phonemes. In the previous study, subjects heard natural CVC prime-target pairs in various degrees of phonetic overlap and named the second item (the target) as quickly as possible. The present study adopted the same approach but utilized a phoneme monitoring task in which subjects responded whenever the target began with a designated phoneme. The pattern of RT results, while different from that obtained in the naming task, again supported a rejection of the phoneme and shed further light on the nature of the representation. Similar patterns across word and nonword blocks and across trials in which voice was the same versus different within a prime-target pair were found with both tasks. A comparison will be made of the two task types in terms of the results generated and the implications for their use in studying on-line spoken word recognition. [Work supported by NINCDS.]
The resolution of modal Doppler shifts in an acoustic waveguide
Authors: Song Hee Chun , Baggeroer Arthur B.
Published: October 2005

A method for estimating the source velocity from the acoustic signal field propagating in a dispersive oceanic waveguide is presented. The signal is observed at an omnidirectional receiver in the presence of additive white noise. The source transmits a continuous-wave (cw) signal, and the source motion is assumed to be uniform (unaccelerated). The ocean is modeled as a waveguide that is horizontally stratified with an arbitrary sound-speed profile in the vertical. The acoustic field generated by a moving point source in terms of normal modes predicts that each mode has a different Doppler shift, which contrasts with a single Doppler shift in a homogeneous unbounded medium. The modes are considered to remain phase locked with each other, i.e., coherent. The source velocity, including the number of modes involved, can be estimated through resolving Doppler shifts. The method is a time-domain interpretation of a recently developed eigenstructure technique for multitarget direction finding with passive antenna arrays, in conjunction with smoothing preprocessing scheme to deal with coherent modes [Shan et al., IEEE Trans. Acoust. Speech Signal Process. ASSP-33 (4) (1985)]. Simulation results that illustrate the performance of the method are presented.
The use of neural networks in acoustic coherence measurements
Authors: Beran Mark J.
Published: October 2005

The intensity distribution of an incoherent target may be determined from coherence measurements. It is shown here how a neural network may be developed to solve this inverse problem. The effect of random phase variations resulting from ocean temperature fluctuations is considered in the formulation.
Analysis of objectionable noises in residential bedrooms
Authors: Wu Me Q. , Gastimeier Bill , Allen D. L.
Published: October 2005

This paper summarizes bedroom noise complaints collected by Vibron in the past 2 years. For the great majority of the 28 cases considered, the noise sources were mechanical equipment. Noise levels in most bedrooms where complaints occurred, were above NC 30. However, about 18% of the complaints came from bedrooms where noise levels were at or below NC 30. The lowest noise level was NC 20. A detailed study of the spectrum and the time signal of the low noise level cases indicates that the spectral and temporal characteristics as well as the background noise level affect the potential for annoyance. Peaks or discontinuities in either frequency or time domain usually cause complaints to noise with lower total levels. Lower background noise levels also increase the noise complaints.
Method to measure ear canal impedance and reflectance
Authors: Keefe Douglas H. , Ling Robert , Bulen Jay C.
Published: October 2005

The accuracy of the single microphone method of measuring impedance depends upon a calibration using a least-mean-squares fit of the Thevenin source characteristics to measured data from a set of closed, cylindrical tubes [Allen, in Peripheral Auditory Mechanisms, edited by Allen et al., pp. 44–51 (1985)]. Using a 2048-point chirp stimulus, the Thevenin parameters are determined at each frequency using the lengths of the closed tubes. These lengths, which depend upon the insertion distance of the foam eartip suitable for use in human ear canals, are optimally chosen using Brent's method. Results on the calibration sensitivity using four versus six closed tubes, of varying lengths and diameters will be presented. The area distance functions of the calibration tubes and ear canal enable calculation of the energy reflectance from impedance data at arbitrary positions in the tubes [Keefe and Ling, J. Acoust. Soc. Am. Suppl. 1 85, S67 (1989)], but the area at the eartip is sufficient to calculate reflectance at the eartip. This area is inferred from acoustic resistance at the eartip, and the accuracy of the inference is limited more by the finite measurement bandwidth than by viscothermal corrections. Results for human adults will be discussed. [Work supported by NIH.]
Nonlinearities of evoked otoacoustic emissions analyzed by means of a deconvolution procedure
Authors: Grandori Ferdinando , Ravazzani Paolo
Published: October 2005

Evoked otoacoustic emissions were measured with clicks and short duration tone bursts at intensities ranging from 0–50 dB HL, in normal hearing subjects. Both the initial acoustic transient and the physiological response were measured and stored. This enables the computation of the deconvolution between cochlear response and input stimulus. In a first experiment, the rationale was the following: Does a simple model (the system's impulse response) explain the input-output relationship with these stimuli? Comparison of the deconvolutions obtained from click-evoked responses with those evoked by tone bursts reveals that, to a good approximation, the impulse response is a reasonable description of the system's dynamical aspects, at a given stimulus intensity. In a second series of experiments, responses and their deconvolutions were analyzed in more detail across the intensity range. Results show that this view is definitely too simplistic; significant deviations from linearity are observed. These deviations are then compared to those of the original otoacoustic emissions, at different post-stimulus times. Of particular interest is the observation that the degree of linearity changes markedly from the early to the late portions of the whole response: This suggests a method for detecting the “true” response.
An effective method for computing the nonlinear equation of modeling basilar membrane motion
Authors: Yang Jun , Fan Changxin
Published: October 2005

Many available methods for the investigation of the cochlea are only concerned with the linearized model; however, the experimental data show that this linearized assumption is not the case. In order to understand better the mechanism by which the cochlea processes speech, it is desirable to investigate as accurately as possible the response characteristics of the cochlea to speech that is reflected by the nonlinear equation. But, the complexity of solving this equation becomes a great obstacle. In this paper, a novel technique for computing the nonlinear equation in a two-dimensional cochlear model is shown. The solution of the nonlinear equation gives the displacement of the basilar membrane (BM). Two steps are adopted for solving the nonlinear equation: The first is to solve the linearized equation related to the BM displacement Z0(x,t), and the second is to seek the difference between the true displacement Z(x,t) of the BM and the obtained Z0(x,t). Summing the difference and Z0(x,t), therefore, the accurate displacement of the BM is obtained. This proposed scheme is easily implemented and is proved to be very effective.
Is the damaged ear more susceptible?
Authors: Zhou Zhiyi , Ward W. Dixon
Published: October 2005

When chinchillas are given daily exposures to noise on Monday through Friday over a period of 9 weeks, the threshold shift at the end of the first week of exposure is at least as great as the shift at the end of the last week. This threshold is a combination of temporary threshold shift (TTS) and permanent threshold shift (PTS). It is generally assumed that the PTS component increases every week. To test this assumption, a week of recovery in quiet was interjected after the first week of exposure to 105-dB, 700- to 2800-Hz noise, 8 h/day, and after the fourth week, so that the PTSs after 1, 4, and 9 weeks of exposure could be compared. They were indistinguishable. The most parsimonious interpretation of this observation is that all of the damage occurred during the first week of exposure, although it could be the result of equal and opposite factors, e.g., some type of “learning” effect that just balances the continued growth of damage postulated by the microtrauma theory of noise-induced hearing loss. In any event, the results prove that a damaged chinchilla ear is not more susceptible to further damage—if anything, it is less susceptible. [Research supported by NS12125.]
Last names are hard to say!
Authors: Calleja Teri , Mniszewski Susan M. , Fox Jennifer E.
Published: October 2005

Are last names harder to pronounce than words in that they do not always follow standard English pronunciation rules? If they are, then names would require special treatment in text-to-speech translation systems. A database was developed of last names and associated pronunciations of Los Alamos National Laboratory employees. An algorithm was developed for scoring the accuracy of a hypothesized pronunciation when compared against ground truth (the employees' own pronunciations of their names). Using the rules in Dectalk, the following sets of data were compared with respect to the accuracy of the rules: names versus words, English names versus non-English names, and non-English names versus words. Significant differences were found among the various conditions. [Work supported by DOE Contract W-7405-ENG-36.]
The CITRON database: Collection procedures and phonetic analysis of customer elicited speech over the telephone network
Authors: Basson Sara , Chigier Benjamin , Jankowski Charles , Spitz Judith , Yashchin Dina
Published: October 2005

The NYNEX artificial intelligence speech technology group is pursuing automation of operator services using speech technology. In order to automate portions of the directory assistance (DA) transaction, speaker-independent recognition of isolated city names is currently under development. In the servioe of this project, a large database of New England city names has been collected. The city name responses over the telephone network (CITRON) database were collected to reflect speaker productions in a goal-oriented man-machine interaction. To collect this database, real customers were presented with digitized prompts requesting the target city name when they called DA. A total of 27 467 calls were collected over a 1-month period. The CITRON database will be characterized along the following dimensions: total number of tokens, duration of tokens, number of phones and syllables per token, stress characteristics of each token, phonetic distribution of tokens, and derivable speaker characteristics. These results will be compared to the distribution of phones in the American English lexicon to characterize differences between the database and the English language.
Automatic labeling of large prosodic databases: Tools and methodology
Authors: Bailly G. , Barbe T. , Veste S. , Wang H. , Tuffelli D.
Published: October 2005

Construction of synthetic prosodic models for high-quality speech synthesis implies the association of linguistic information and prosodic information on a corpus especially designed to study certain linguistic insights [Emerard and Benoit, 16èmes Journées d'Etudes sur la Parole, Hammamet, Tunisie, 1987, pp. 224–226]. This paper concerns the automatic acquisition of quantitative description of the prosodic contours compatible with a high-quality synthesis approach. Originality of this approach is that the sound continuum's description is based on a phonetic model, i.e., phonemes as emergence functions issued by a temporal decomposition (TD) technique [Bailly, Marteau, and Abry, Proc. Int. Conf. ASSP, Glasgow, Scotland, 1989, pp. 508–511]. The speaker-independent automatic alignment [Wang et al., J. Acoust. Soc. Am. Suppl. 1 87, S106 (1990)] and F0 tracker are presented: They deliver a prosodic contour description (duration, nucleus duration, F0 contour, energy). This description is sufficient to describe high-quality prosodic contours and can then be used to test the quality of synthetic rule-generated contours: Prosodic models can also be tested using the overlap and add (OLA) pitch-synchronous method [Charpentier and Stella, Proc. Int. Conf. ASSP, Tokyo, Japan, 1986, pp. 2015–2018]. The above pitch tracker generates pitch marks automatically, which are used to distort the original natural data in order to generate a high-quality synthetic signal with the desired prosodic contour. This methodology (entirely automatic implemented) enables a test of generatire prosodic models on the original data as well as test data of the segmental problems getting read. Diphone can therefore be labeled with the same procedure and a complete synthesis system can also be built.
On the use of F0 variations in automatic speech recognition
Authors: Lefèbvre Claude , Zwierzyński Dariusz A.
Published: October 2005

This study deals with the feasibility of using microprosodic pitch variations in CVC words in automatic speech recognition. It has been demonstrated in the past [K. Silverman, Proc. Inst. Acoust. 6, 445–452 (1984)] that the fundamental frequency (F0) contour at a post-release voicing onset of CVC words can serve as a cue in distinguishing between voiced and voiceless consonants. The word pitch contours in the CVC speech database obtained from MIT were examined. The word-initial consonants in this study involved stops /p, t, k, b, d, g/ and affricates /tš, dž/, whereas the vowels included /i, ᴜ, æ/. The algorithm that was used in pitch measurement is an average magnitude difference function (AMDF). With this method, pitch is estimated by running a half-wave rectified LPC residual of the speech waveform through the AMDF. Pitch measurements confirm that, for a majority of CVC words in this database, fundamental frequency in the initial glottal cycles of a post-release vowel is higher for voiceless consonants than for voiced ones. Post-onset F0 follows a rising or falling trajectory into the vowel, depending on the global F0 pattern in the word. These tests indicate that F0 post-release onset values may provide more reliable information than the direction of the F0 contour in the vowel for making a voiced—voiceless decision in the automatic speech recognition system [M. J. Hunt and C. Lefèbvre, ICASSP-89, 21.S6.3 (1989)]. [Work supported by NSERC Operating Grant C48 6681.]
A fast tree-trellis search for finding the N-best sentence hypotheses in continuous speech recognition
Authors: Soong Frank K. , Huang Eng-Fong
Published: October 2005

In this paper a tree—trellis-based fast search for finding the N-best sentence hypotheses in continuous speech recognition is proposed. The search consists of two parts: a forward, time-synchronous, trellis search and a backward, time asynchronous, tree search. In the first module the well-known Viterbi algorithm is used for preparing a map of all partial paths and the corresponding scores, time synchronously. In the second module a tree search is used to grow partial paths backward and time asynchronously. Each partial path in the backward tree search is ranked ordered in a stack by the corresponding full path score, which is computed by adding the partial path score with the best possible score of the remaining path obtained from the trellis path map. In each path growing cycle, the current best partial path, which is at the top of the stack, is extended by one arc (word). The new tree-trellis search is different from the traditional time synchronous Viterbi search in its ability for finding not just the best but the N-best paths of different word content. The new search is also different from the A* algorithm, or the stack algorithm, in its capability for providing an exact, full path score estimate of any given partial (i.e., incomplete) path before its completion. When compared with the best candidate Viterbi search, the search complexities for finding the N-best strings are rather low, i.e., only a fractional increase in computation is needed. Also, only a marginal increase of memory space is required for implementing the tree-trellis algorithm. This new algorithm provides a natural interface between a continuous speech recognizer and its higher level processing modules of a speech understanding system.
Automatic segmentation and alignment of continuous speech based on the temporal decomposition model
Authors: Wang H. D. , Bailly G. , Tuffelli D.
Published: October 2005

A robust and speaker-independent algorithm for the automatic segmentation of speech has been designed. It aligns a phonetic transcription with a phoneme nucleus detector, which is based on the temporal decomposition paradigm (TD) [B. Atal, IEEE Trans. Acoust. Speech Signal Process. ASSP-26, 81–84 (1983); Bailly, Marteau, and Abry, Proc. Int. Conf. ASSP, Glasgow, Scotland, 508–511 (1989)]: The phonetic string is seen as overlapping emergence functions (EFs) whose maxima arise for the phoneme nucleus. The segmenter minimizes the reconstruction error (least-squares error) between the time-frequency representation of the speech signal and the above model. The automatic segmentation and alignment algorithm performs the task in three steps: (a) predetection of phonemes nuclei centers, (b) time alignment of the corresponding phonetic transcription, and (c) adjustment of these output nuclei centers and phoneme boundaries detection. The first step, which is inspired by Van Hemert's work [Van Hemert, Philips Tech. Rev. 43(9), 233–242 (1987)], uses an adaptive detection window to produce phonemes nuclei centers candidates. The second step uses the dynamic time warping (DTW) procedure to align these candidates with the known corresponding phonetic transcription. This DTW is guided by anchor points: a crude local probability function takes account of energy and zero-crossings distributions for each phoneme. A new temporal decomposition technique gives an analytical solution with fixed number of targets and no compacity constraints. The TD errors between three consecutive candidates are used to calculate transient costs, thus enabling insertion and omission of nuclei centers. The third step adjusts the nuclei centers on the center of gravity of each of the corresponding EFs. It also produces a phoneme boundary segmentation considered by the time of equal adjacent EFs. This algorithm has been trained using 200 sentences pronounced by one speaker and tested using 50 sentences pronounced by seven speakers. On the test corpus, 86% of the phonemes nuclei centers candidates fall alone into one manual segment. In addition, 94% of the final nuclei centers match the manual segmentation.
Speech processing using an auditory model and neural networks
Authors: Anderson Timothy R.
Published: October 2005

A neural network that employs unsupervised learning was used on the output of a neurophysiologically based model of the auditory periphery [K. L. Payton, J. Acoust. Soc. Am. 83, 145–162 (1988)] to perform phoneme recognition. The auditory model incorporates steps describing the conversion from the acoustic pressure-wave signal at the eardrum to the time course activity in auditory neurons. The model can process arbitrary time domain waveforms and yields the predicated neural firing rate. Continuous speech from ten speakers (three female and seven male) taken from the TIMIT database [W. Fisher et al., J. Acoust. Soc. Am. Suppl. 1 81, S92 (1987)] was processed through the auditory model for this experiment. The average firing rate of 20 channels from the auditory model was used to train a Kohonen self-organizing feature map. The resulting context-independent phoneme recognition performance (30% correct) was comparable to that of the SPHINX System [K. F. Lee and H. W. Hon, IEEE Trans. Acoust. Speech Signal Process. ASSP-37, 1641–1648 (1989)] with the same number of speakers in the training set and no smoothing of parameters.
Phonetic priming in spoken word recognition: Task comparisons
Authors: Gagnon Deborah A.
Published: October 2005

In an effort to reveal the nature of the representation underlying spoken word recognition, a basic priming paradigm has been used across a series of different tasks. The paradigm involves presenting subjects with natural CVC prime-target pairs in various degrees of phonetic overlap, and asking the subjects to respond to the target as fast as possible in a task-specific manner. Previous reports on the use of this paradigm described the results from a naming task [Gagnon and Sawusch, J. Acoust. Soc. Am. Suppl. 1 85, S125 (1989)] and a phoneme monitoring task [Gagnon and Sawusch, J. Acoust. Soc. Am. Suppl. 1 86, S99 (1989)]. The pattern of results obtained in the naming task reflected a position-specific phonetic representation, while the phoneme monitoring task yielded results supportive of an allophonic representation. The methodology has been extended to a lexical decision task in which subjects determine whether the target item is a word or nonword. A comparison of the results obtained from the lexical decision task to those obtained from the naming and phoneme monitoring tasks will be made, and conclusions regarding the appropriateness of these tasks for investigating the representation used in spoken word recognition will be offered. [Work supported by NIDCD Grant DC00219 to SUNY at Buffalo.]
Talker variability and word recognition: A developmental perspective
Authors: Oliver Brigette R.
Published: October 2005

The effects of talker variability on spoken word recognition were studied developmentally in 3-, 4-, and 5-yr-old children. Subjects listened to lists of words and were required to identify each word by pointing to a picture in a six alternative visual display. The words and pictures were taken from the word intelligibility by picture identification test (WIPI). Three talker conditions were examined: single talker, single talker with varying amplitude, and multiple talker. Each child heard three lists, one from each talker condition, in varying order. Results showed a main effect of age and an interaction between talker condition and list order. As expected, we found an increase in overall accuracy with age. However, the 3-yr-olds did better in the single-talker condition than the other conditions (which did not differ). This was found only when the lists were presented first during a test session. Second and third lists showed no difference due to talker condition. In most of the conditions the 4- and 5-yr-olds displayed ceiling levels of performance that may have obscured other differences. Further studies are underway with younger children to examine the nature of the differences in perception between single-talker and multi-talker lists. [Work supported by NIH Grant DC-0111-14 to Indiana Univ.]
On the role of amplitude and phase in the synthesis of male and female voices
Authors: Verhelst W. D. E. , Eggen J. H.
Published: October 2005

A pitch-synchronous segmentation, which was shown to be perceptually close to a deconvolution [C. Hamon et al., Proc. IEEE ICASSP'89, 238–241 (1989)], was used to obtain a short-time Fourier representation of the LPC residual. After selected amplitude and phase manipulations of voiced segments, a residue was reconstructed, which was used to drive the LPC synthesis filter. Twenty utterances (ten male, ten female) were investigated under two amplitude (original/flat) and two phase conditions (original/zero), yielding four versions for each utterance. The quality of these versions was judged by 12 subjects in a paired-comparison experiment. Original amplitude information was consistently preferred over original phase information. For female voices, there were significant quality differences between any of the four versions. However, for male voices the original amplitude information alone proved to be sufficient to make the synthetic speech almost indistinguishable from natural speech. [Work was supported in part by the Dutch SPIN-ASSP program.]
Speech preprocessing using analog VLSI
Authors: Liu Weimin , Andreou Andreas G. , Goldstein Moise H.
Published: October 2005

The aim of this project is analog CMOS VLSI implementation of a low-power, real-time front-end processor for speech recognition and for multichannel aids for the deaf. Physiological studies of the mammalian auditory periphery indicate that synchrony coding of speech signals by the temporal pattern of activity in the auditory nerve is noise resistant and performs amplitude compression without distortion over a wide range [M. B. Sachs, H. F. Voigt, and E. D. Young, J. Neurophysiol. 50, 27–45 (1983)]. In a number of studies using digital computation to model the auditory periphery, the properties of synchrony coding of speech signals have been explored [K. L. Payton, J. Acoust. Soc. Am. 83, 145–162 (1988), and articles in J. Phonet. 16, 1 (1988)]. Here VLSI realization of the middle ear, the basilar membrane, and the hair cell and synapse are presented along with design features of the analog VLSI approach. The multichannel outputs correspond to the time-varying firing rates of neurons from discrete places on the cochlear partition.
A study of acoustical enhancement systems, leading to the use of time-variant synthetic reverbaretion—Part II
Authors: Griesinger David , Muncy Neil
Published: October 2005

The previous paper described the theoretical background of a new system developed for the Elgin Theater in Toronto. This paper goes further into the theoretical advantages of the system and deals with its practical realization. In brief, it comprises two or more microphones placed 25 to 50 ft from the sound sources, an array of 120 loudspeakers arranged in eight interleaved banks, and digital electronics with 16 independent time-variant reverberators. It will also describe, and if possible demonstrate, the results obtained, illustrating the very successful illusion of normal reverberation and its great flexibility.
Room-average absorption coefficient determination by means of intesity measurements in steady-state acoustic field conditions
Authors: Stanzial Domenico , Fuschini Augusto
Published: October 2005

Absorption coefficients are currently indirectly obtained from reverberation time measurements, involving impulsive or transient sound phenomena. The here-proposed method, based on steady sound conditions, is complementary to the current one and allows one to obtain the full frequency spectrum of absorption coefficients from intensity measurements. The method is based on a model of room acoustics in terms of a linear-feedbacked system. Following the system theory approach, the room-average absorption coefficient is readily recognized as 1 − H, where H is the feedback transfer function. As intensity measurements provide experimental evaluation of the room transfer function, which is simply related to the feedback transfer function; absorption coefficients are consequently determined.
Interactive computer tools for image method room modeling
Authors: Kovitz Paul S. , Swanson David
Published: October 2005

The image method for determining the source to receiver transfer function in an enclosure has been implemented on computers for some time now [J. Allen and D. Berkley, J. Acoust. Soc. Am. 65, 943 (1979); J. Borish, J. Acoust. Soc. Am. 75, 1827 (1984)]. However, it is sometimes difficult to apply the results of these models in terms of, for example, spatial hearing phenomena, simply by looking at the time domain impulse response. An interactive computer program that displays, in three dimensions, the source, the receiver, the enclosure, and several representations of the sound rays incident on the receiver, will be demonstrated. A real-time stereo DSP system ,that processes sound with the simulated impulse response, so the results of the simulation may be heard, will also be demonstrated. Comparisons of the simulated results with experimental results will be available.
Fourth-order spectra of intensity-modulated sinusoids
Authors: Dwyer Roger F.
Published: October 2005

Often propagating underwater signals are intensity modulated by inhomogeneities in the medium. Since this is a multiplicative phenomenon the resulting spectrum is a convolution between the modulation and the desired signal. However, it has been recently shown that a special case of the fourth-order cumulant spectrum can extract the frequency of an intensity-modulated sinusoid without distortion from convolution. The special case of the fourth-order cumulant spectrum is also independent of the covariance of the modulation and only depends on the variance. In order to test the theoretical results an experiment was conducted. An amplitude-modulated sinusoid was transmitted through the water and received. Both the spectrum and the fourth-order cumulant spectrum were estimated from the received data. The experiment corroborated the theoretical results.
Fourier synthesis of the scattered pressure field from ridgelike features using two-dimensional boundary integral equation methods
Authors: Fawcett John A. , Dawson Trevor W.
Published: October 2005

In earlier work an integral equation method for the computation of the pressure field scattered from deformations of a two-dimensional oceanic waveguide was presented. The boundary conditions on the top and bottom surfaces of the waveguide were taken to be either free or rigid. In this presentation the three-dimensional scattered field from ridgelike features is computed by solving a sequence of two-dimensional problems for different values of the horizontal wavenumber and then using Fourier synthesis to construct the three-dimensional response. In a manner analogous to fast-field propagation models, the Fourier inversion is performed along a contour in complex-wavenumber space. As well, a simple stationary phase expression is derived for the pressure scattered from a ridge in the farfield. Numerical examples are given.
Horizontal directionality of the ocean infrasonic sound field: The effect of bottom topography
Authors: D'Spain G. L. , Hodgkiss W. S. , Edmonds G. L.
Published: October 2005

Simultaneous measurements of the acoustic pressure and particle velocity in the infrasonic (1–20-Hz) band are made by independent, freely drifting, neutrally buoyant Swallow floats. Horizontal directionality of the ambient sound held as determined by “DIFAR” beamforming is compared to that determined by the horizontal projection of the active acoustic intensity vector. Results from an experiment 300 km west of San Diego in 3.8-km-deep water indicate dominant acoustic energy flow from the southwest between 4 and 12 Hz, even though a storm occurred in the Gulf of Alaska (to the northwest) and the Southern California shipping lanes exist to the east-northeast. The southwest direction coincides with the center of a window created by the ocean bottom topography. Rudimentary numerical modeling using the SACLANTCEN parabolic equation code PAREQ is used to help explain these results. Above 16 Hz, the directionality is determined by whale noise. [Work supported by ONR.]
Probing porous media with superfluid acoustics
Authors: Linton David
Published: October 2005

Those properties of porous media that can be deduced from experiments using measurements of superfluid 1st, 2nd, 4th, and 3rd sound are discussed; the transferability of these results to other transport experiments, especially the acoustic properties of porous media saturated with Newtonian fluids, is also explored. Many of the relevant geometrical parameters are those that arise in a canonical electrical conductivity problem in which the porous solid is insulating, the pore fluid is conducting, and there is an additional surface conductivity lining the walls of the pore space. The most important geometrical parameters are the three-dimensional tortuosity of the pore space, α2 the two-dimensional tortuosity of the pore/grain interface, α2, and ʌ, which is a well-defined measure of dynamically connected pore sizes.
Sound propagation in superfluid helium as a probe of the microscopic geometry of a porous solid
Authors: Baker S. R.
Published: October 2005

Liquid helium is unique among substances on Earth in that when it is cooled to extremely low temperatures it has the ability to flow without friction through the tiniest of channels. More than any other, this property of liquid helium at low temperatures, or superfluid helium, as it is known, makes sound propagation in it a superior probe of the microscopic geometry of a porous solid. It will be shown how the three Blot parameters that characterize the microscopic geometry of a porous solid may be extracted from speed and attenuation measurements of the propagating sound modes in superfluid helium contained within it. The results of experiments in which such measurements were made for a variety of porous solid samples will be described. [Work supported by ONR, NRL-USRD, and the Naval Postgraduate School.]
Finite-difference modeling of acoustic waveforms in a fluid-filled borehole surrounded by a Biot porous media
Authors: Kostek Sergio , Bayliss Alvin
Published: October 2005

Analytical techniques are very efficient in dealing with either homogeneous or layered fluid-saturated poroelastic media. Inhomogeneities in the material properties or complicated interfacial boundaries make the problem analytically intractable, forcing then the use of numerical techniques such as finite-difference, finite element, etc. An explicit finite-difference time domain modeling scheme that is able to handle fluid, elastic, and fluid-saturated poroelastic complex media is presented. In particular, an axisymmetric version of the code is used to model acoustic wave propagation in fluid-filled boreholes surrounded by a Biot type of medium. The effects of borehole rugosity, mudcake, and vertical layering are investigated. Comparisons with analytical solutions in simple cases are also made, showing good agreement between the different methods. Absorbing boundary conditions for application in the porous regions were also developed and proved to be effective in avoiding spurious reflections.
Forced response and acoustical radiation for beams with periodic and quasiperiodic attachments
Authors: Yuang Ten-Bin , Pate Anna L.
Published: October 2005

A uniform Bernouli-Euler beam with multiple, concentrated mass/stiffness/damping attachments is investigated with the transfer matrix method. The method is similar to that of Y. K. Lin and T. J. McDaniel [Trans. ASME, J. Eng. Ind. (Nov. 1969)]. The response of the beam to a point force as well as the associated acoustical radiation are calculated. Specifically, the effect of periodic and quasiperiodic attachments on the beam response is of interest. Numerical results are compared with the results of experimental studies of both vibrations and nearfield and farfield acoustical radiation of the beam. [Work supported by ONR.]
Experiments on linear and nonlinear wave propagation in random and quasiperiodic media
Authors: McKenna Mark J. , Spoor P. S. , Stanley R. L. , Dimasi Elaine , Maynard J. D.
Published: October 2005

Since the development of Anderson localization and the discovery of aluminum alloy quasicrystals, there has been considerable experimental and theoretical interest in wave propagation in random and quasiperiodic systems. More recently, there has been interest in the nonlinear properties of such systems. Having completed some experimental research on the linear behavior of waves in a one-dimensional random system and a two-dimensional quasiperiodic system, the nonlinear properties of such systems are now being studied. Two systems, one having a local nonlinear interaction (surface waves in a superfluid film) and the other having a global nonlinear interaction (transverse waves in a stretched string), will be described. [Work supported by NSF DMR 8701682 and the Office of Naval Research.]
Linear and nonlinear acoustic propagation in a periodic waveguide
Authors: Bradley Charles E.
Published: October 2005

The effect of periodic nonuniformity of a waveguide on the propagation of both infinitesimal and finite-amplitude waves is investigated theoretically and experimentally. Analytic expressions for a dispersion relation, the impedance function, and the Bloch wavefunction are derived for the case of linear, plane wave mode propagation in a rectangular wave-guide that is periodically loaded with rigidly terminated side branches. Experiments have been done in a 25.4-mm × 38.1-mm × 6-m air-filled aluminum waveguide with 38.l-mm-deep side branches at 0.l-m intervals. Measurements verify the predicted passband/stopband structure of the dispersion relation and the forward and backward traveling wave composition of the Bloch wavefunction. In the case of finite-amplitude excitation, the compound wave composition of the fundamental Bloch wave results in a bidirectional excitation of the second harmonic. Preliminary measurements show that second harmonic behavior is qualitatively similar to that for an ordinary dispersive medium in that parametric upconversion is effectively blocked. Possible application in the amplification of traveling waves is discussed. [Work supported by U.S. Office of Naval Research.]
Tissue motion assessment from scattered ultrasound echoes
Authors: Hein Ilmar A. , O'Brien William D.
Published: October 2005

Ultrasound has become a standard clinical tool in the diagnosis and treatment of illness and injury. Perhaps the most well-known use of ultrasound is the noninvasive imaging of different parts of the body. Ultrasound does, however, have other applications in medical diagnoses, such as in tissue characterization and in the measurement of tissue motion in vivo. The measurement of tissue motion is a somewhat broad category. It can include the motion of physically active organs such as the heart and discrete structures such as cardiac valves and arterial walls to diagnose heart disease. It can be used to measure motion in soft tissues (such as the liver) due to the pulsing of an artery in the tissue. The soft tissue motion is a function of elasticity of the tissue, which is directly related to the state of the tissue (diseased or healthy). Finally, the measurement of blood flow velocity is an important parameter in diagnosing coronary diseases such as venous thrombosis. Regardless of the particular application of ultrasound, all applications require the transmission and capture of rf ultrasonic signals. After capture, the signals must be processed in some way to extract the desired information. In the case of estimating tissue motion, many methods exist, ranging from the well-known and widely implemented Doppler technique to newly developed time domain methods; from estimating velocity from a single rf A-scan to using multiple M-mode video images. The topic of this presentation is to review and compare some of the methods currently under development to measure tissue motion from scattered ultrasound echoes. [Work supported by NIH/HL 39704 and NIH/CA 36029.]
Analysis and computations of measurement system effects in ultrasonic scattering experiments
Authors: Waag Robert C. , Astheimer Jeffrey P. , Smith James F.
Published: October 2005

A model that characterizes the effects of beams and waveforms on the measurement of ultrasonic scattering is described and analyzed. The model includes arbitrary emitter beam patterns and detector apertures as angular spectra of plane waves. Arbitrary emitted pulses and detector time gates are incorporated through frequency spectra of temporal harmonics. A transformation of variables is employed to express the spectrum of the measured pressure as a product in wave space of a system function and the Fourier transform of the medium variations. The analysis shows the influence of beam patterns and a finite detector gate, relates the current model to a prior narrowband analysis, provides asymptotic expressions for beam pattern effects produced by radially symmetric Gaussian apertures, and treats the backscattering configuration for arbitrary apertures and correlated scattering. Numerical computations illustrate the effect of Gaussian, exponential, and uniform circularly symmetric spatial aperture functions on the measurement of the power spectrum of medium variations. The results may be used to design scattering experiments in which measurements yield scaled values of intrinsic parameters of scattering.
Mixture laws for ultrasonic tissue characterization
Authors: Jiang Peng , Apfel Robert E. , Everbach E. Carr
Published: October 2005

Five mixture laws for predicting tissue composition are studied. They are (1) the density law, ρ = Σi Xi ρi; (2) the sound speed law, 1/c = Σi X1(1/ci); (3) the compressibility law, β = Σi, Xi, βi; (4) the phase shift parameter law, (B/4)/pc3 = Σi, Xi[(B/A)/pc3]i; and (5) the nonlinear parameter law, β2(1 + B/2A) = Σi Xi, [Β2(1 + B/2A)]i; where ρ is the density of the mixture, c is the sound speed of the mixture, Β is the compressibility of the mixture, B/A is the nonlinear parameter of the mixture, and Xi is the volume fraction of the ith component of the mixture. The subscripted parameters correspond to the parameters of the components of the mixture. Based on these experiments, it is shown that some combinations of the mixture laws, under the constraint condition Σi, = 1, can provide a good prediction of tissue composition. Both theoretical and experimental results will be presented. The relations among the mixture laws will be outlined and limitations of the laws will be discussed. [Work supported by the National Institutes of Health through Grant 2ROlGM30419.]