Showing papers in "Journal of the Acoustical Society of America in 1995"
TL;DR: In this paper, an acoustic imaging system for use within a heart has a catheter, an ultrasound device, and an electrode mounted on the catheter to create an ultrasonic image, and the electrode is arranged for electrical contact with the internal structure.
Abstract: An acoustic imaging system for use within a heart has a catheter, an ultrasound device incorporated into the catheter, and an electrode mounted on the catheter. The ultrasound device directs ultrasonic signals toward an internal structure in the heart to create an ultrasonic image, and the electrode is arranged for electrical contact with the internal structure. A chemical ablation device mounted on the catheter ablates at least a portion of the internal structure by delivery of fluid to the internal structure. The ablation device includes a material that vibrates in response to electrical excitation, the ablation being at least assisted by vibration of the material. The ablation device may alternatively be a transducer incorporated into the catheter, arranged to convert electrical signals into radiation and to direct the radiation toward the internal structure. The electrode may be a sonolucent structure incorporated into the catheter, through which the ultrasound device is arranged to direct signals. An acoustic marker mounted on the catheter emits a sonic wave when electrically excited. A central processing unit creates a graphical representation of the internal structure, and super-imposes items of data onto the graphical representation at locations that represent the respective plurality of locations within the internal structure corresponding to the plurality of items of data. A display system displays the graphical representation onto which the plurality of items of data are super-imposed.
1,541 citations
TL;DR: Differences between the psychometric functions for high- and low-context conditions were used to show that both groups of old listeners derived more benefit from supportive context than did young listeners, and supporting a processing model in which reallocable processing resources are used to support auditory processing when listening becomes difficult either because of noise, or because of age-related deterioration in the auditory system.
Abstract: Two experiments using the materials of the Revised Speech Perception in Noise (SPIN-R) Test [Bilger et al., J. Speech Hear. Res. 27, 32-48 (1984)] were conducted to investigate age-related differences in the identification and the recall of sentence-final words heard in a babble background. In experiment 1, the level of the babble was varied to determine psychometric functions (percent correct word identification as a function of S/N ratio) for presbycusics, old adults with near-normal hearing, and young normal-hearing adults, when the sentence-final words were either predictable (high context) or unpredictable (low context). Differences between the psychometric functions for high- and low-context conditions were used to show that both groups of old listeners derived more benefit from supportive context than did young listeners. In experiment 2, a working memory task [Daneman and Carpenter, J. Verb. Learn. Verb. Behav. 19, 450-466 (1980)] was added to the SPIN task for young and old adults. Specifically, after listening to and identifying the sentence-final words for a block of n sentences, the subjects were asked to recall the last n words that they had identified. Old subjects recalled fewer of the items they had perceived than did young subjects in all S/N conditions, even though there was no difference in the recall ability of the two age groups when sentences were read. Furthermore, the number of items recalled by both age groups was reduced in adverse S/N conditions. The resutls were interpreted as supporting a processing model in which reallocable processing resources are used to support auditory processing when listening becomes difficult either because of noise, or because of age-related deterioration in the auditory system. Because of this reallocation, these resources are unavailable to more central cognitive processes such as the storage and retrieval functions of working memory, so that "upstream" processing of auditory information is adversely affected.
1,030 citations
TL;DR: Foreign accents were evident in sentences spoken by many NI subjects who had begun learning English long before what is traditionally considered to be the end of a critical period, and Gender was also found to influence degree of foreign accent.
Abstract: This study assessed the relation between non‐native subjects’ age of learning (AOL) English and the overall degree of perceived foreign accent in their production of English sentences. The 240 native Italian (NI) subjects examined had begun learning English in Canada between the ages of 2 and 23 yr, and had lived in Canada for an average of 32 yr. Native English‐speaking listeners used a continuous scale to rate sentences spoken by the NI subjects and by subjects in a native English comparison group. Estimates of the AOL of onset of foreign accents varied across the ten listeners who rated the sentences, ranging from 3.1 to 11.6 yr (M=7.4). Foreign accents were evident in sentences spoken by many NI subjects who had begun learning English long before what is traditionally considered to be the end of a critical period. Very few NI subjects who began learning English after the age of 15 yr received ratings that fell within the native English range. Principal components analyses of the NI subjects’ responses to a language background questionnaire were followed by multiple‐regression analyses. AOL accounted for an average of 59% of variance in the foreign accent ratings. Language use factors accounted for an additional 15% of variance. Gender was also found to influence degree of foreign accent.
817 citations
TL;DR: In this paper, a Rayleigh-Plesset-like equation describing the dynamics of surface-contaminated gas bubbles is derived, which predicts that the surface layer supports a strain that counters the Laplace pressure and stabilizes the bubble against dissolution.
Abstract: Most previous theoretical investigations of gas bubble dynamics have assumed an uncontaminated gas–liquid interface. Recently, however, the potential importance of layers of surface active agents on bubble dynamics has been increasingly recognized. In this work it is assumed that a continuous layer of incompressible, solid elastic material separates the gas from the bulk Newtonian liquid. Elasticity is modeled to include viscous damping. A Rayleigh–Plesset‐like equation describing the dynamics of such surface‐contaminated gas bubbles is derived. The equation predicts that the surface layer supports a strain that counters the Laplace pressure and thereby stabilizes the bubble against dissolution. An analytical solution to this equation which includes both the fundamental and second‐harmonic response is presented. The dispersion relation describing the propagation of linear pressure waves in liquids containing suspensions of these bubbles also is presented. It is found that (1) the resonance frequencies of ...
657 citations
TL;DR: A software package with a modular architecture has been developed to support perceptual modeling of the fine-grain spectro-temporal information observed in the auditory nerve, including new forms of periodicity-sensitive temporal integration that generate stabilized auditory images.
Abstract: A software package with a modular architecture has been developed to support perceptual modeling of the fine‐grain spectro‐temporal information observed in the auditory nerve. The package contains both functional and physiological modules to simulate auditory spectral analysis, neural encoding, and temporal integration, including new forms of periodicity‐sensitive temporal integration that generate stabilized auditory images. Combinations of the modules enable the user to approximate a wide variety of existing, time‐domain, auditory models. Sequences of auditory images can be replayed to produce cartoons of auditory perceptions that illustrate the dynamic response of the auditory system to everyday sounds.
594 citations
583 citations
TL;DR: In this article, the role of atmospheric pressure is emphasized by giving formulas in which the absorption, frequency, and relative humidity are all scaled with respect to atmospheric pressure, and new, more readable and useful figures showing atmospheric absorption as a function of frequency, relative humidity, and atmospheric pressure are presented.
Abstract: This Letter is an extension of an earlier Letter by Bass et al., ‘‘Atmospheric absorption of sound: Update’’ [J. Acoust. Soc. Am. 88, 2019–2021 (1990)]. Errors in a formula for saturation vapor pressure are corrected, and an alternative, much simpler formula is given. The role of atmospheric pressure is emphasized by giving formulas in which the absorption, frequency, and relative humidity are all scaled with respect to atmospheric pressure. Also presented are new, more readable and useful figures showing atmospheric absorption as a function of frequency, relative humidity, and atmospheric pressure. The new figures make it possible to estimate accurately the absorption at any value of atmospheric pressure.
412 citations
TL;DR: Simulations show reasonable similarity to observed vocal-fold motion, measured vertical phase difference, and mucosal wave velocity, as well as experimentally obtained intraglottal pressure.
Abstract: A simple, low-dimensional model of the body-cover vocal-fold structure is proposed as a research tool to study both normal and pathological vocal-fold vibration. It maintains the simplicity of a two-mass model but allows for physiologically relevant adjustments and separate vibration of the body and the cover. The classic two-mass model of the vocal folds [K. Ishizaka and J. L. Flanagan, Bell Syst. Tech. J. 51, 1233-1268 (1972)] has been extended to a three-mass model in order to more realistically represent the body-cover vocal-fold structure [M. Hirano, Folia Phoniar. 26, 89-94 (1974)]. The model consists of two "cover" masses coupled laterally to a "body" mass by nonlinear springs and viscous damping elements. The body mass, which represents muscle tissue, is further coupled laterally to a rigid wall (assumed to represent the thyroid cartilage) by a nonlinear spring and a damping element. The two cover springs are intended to represent the elastic properties of the epithelium and the lamina propria while the body spring simulates the tension produced by contraction of the thyroarytenoid muscle. Thus contractions of the cricothyroid and thyroarytenoid muscles are incorporated in the values used for the stiffness parameters of the body and cover springs. Additionally, the two cover masses are coupled to each other through a linear spring which can represent vertical mucosal wave propagation. Simulations show reasonable similarity to observed vocal-fold motion, measured vertical phase difference, and mucosal wave velocity, as well as experimentally obtained intraglottal pressure.
407 citations
TL;DR: Ear-canal measurements are related to cochlear mechanics by assuming that the transfer characteristics of the middle ear vary slowly with frequency compared to oscillations in the emission spectrum, and Measurements of basilar-membrane motion in the squirrel monkey are used to predict the spectral characteristics of their emissions.
Abstract: Current models of evoked otoacoustic emissions explain the striking periodicity in their frequency spectra by suggesting that it originates through the reflection of forward‐traveling waves by a corresponding spatial corrugation in the mechanics of the cochlea. Although measurements of primate cochlear anatomy find no such corrugation, they do indicate a considerable irregularity in the arrangement of outer hair cells. It is suggested that evoked emissions originate through a novel reflection mechanism, representing an analogue of Bragg scattering in nonuniform, disordered media. Forward‐traveling waves reflect off random irregularities in the micromechanics of the organ of Corti. The tall, broad peak of the traveling wave defines a localized region of coherent reflection that sweeps along the organ of Corti as the frequency is varied monotonically. Coherent scattering occurs off irregularities within the peak with spatial period equal to half the wavelength of the traveling wave. The phase of the net ref...
393 citations
TL;DR: A High Intensity Focused Ultrasound system is provided for treatment of focal disease and the preferred embodiment includes an intracavity probe having two active ultrasound radiating surfaces with different focal geometries.
Abstract: A High Intensity Focused Ultrasound system is provided for treatment of focal disease. The preferred embodiment includes an intracavity probe having two active ultrasound radiating surfaces with different focal geometries. Selectively energizing the first surface focuses therapeutic energy a first distance from the housing, energizing the second surface focuses therapeutic energy nearer the housing. Preferably, the probe includes a thin, flexible, inelastic membrane which is rigidized with pressure to allow blunt manipulation of tissue. Methods of use of the probe are also provided, particularly for treatment of diseases of the prostate.
374 citations
TL;DR: In this article, the results of an articulatory investigation of the supraglottal correlates of linguistic prominence in English are reported, and a proposal of a unified description of linguistic stress is reported.
Abstract: The results of an articulatory investigation of the supraglottal correlates of linguistic prominence in English, and a proposal of a unified description of linguistic stress are reported. Three models of stress are evaluated: that prominence expands jaw movement, that stress expands an abstract articulatory scale involving the opening and closing of the vocal tract, and that stress involves a localized shift toward hyperarticulate speech. A corpus of x‐ray microbeam records of sensible speech is studied, within which the stress pattern is controlled and is checked by means of an intonational analysis. Jaw movement data yield similar results to earlier studies, but kinematic differences interpreted with reference to a gestural theory suggest that different subjects use different articulatory strategies to articulate stress contrasts. In addition, the jaw, lip, and tongue interact in the articulation of stress in subject dependent ways. Thus the articulation of stress should be formulated in terms of abstract articulatory goals, rather than in terms of individual articulator positioning. Finally, the data show that stress affects the articulation of nonsonority distinctions such as backness in vowels and point of articulation in consonants. A hyperarticulation model of stress is discussed in terms of these results.
TL;DR: In this article, the use of a numerical time-domain simulation based on the finite-difference timedomain approximation for studying low and middle-frequency room acoustic problems is described, and an interesting approach lies in using the FDTD simulation to adapt a digital filter to represent the acoustical transfer function from source to observer.
Abstract: This paper illustrates the use of a numerical time‐domain simulation based on the finite‐difference time‐domain (FDTD) approximation for studying low‐ and middle‐frequency room acoustic problems. As a direct time‐domain simulation, suitable for large modeling regions, the technique seems a good ‘‘brute force’’ approach for solving room acoustic problems. Some attention is paid in this paper to a few of the key problems involved in applying FDTD: frequency‐dependent boundary conditions, non‐Cartesian grids, and numerical error. Possible applications are illustrated with an example. An interesting approach lies in using the FDTD simulation to adapt a digital filter to represent the acoustical transfer function from source to observer, as accurately as possible. The approximate digital filter can be used for auralization experiments.
TL;DR: Results from the largest and longest longitudinal study reported to date of changes in pure-tone hearing thresholds in men and women screened for otological disorders and noise-induced hearing loss show gender differences in hearing levels and show that age-associated hearing loss occurs even in a group with relatively low-noise occupations.
Abstract: Current studies are inconclusive regarding specific patterns of gender differences in age‐associated hearing loss. This paper presents results from the largest and longest longitudinal study reported to date of changes in pure‐tone hearing thresholds in men and women screened for otological disorders and noise‐induced hearing loss. Since 1965, the Baltimore Longitudinal Study of Aging has collected hearing thresholds from 500 to 8000 Hz using a pulsed‐tone tracking procedure. Mixed‐effects regression models were used to estimate longitudinal patterns of change in hearing thresholds in 681 men and 416 women with no evidence of otological disease, unilateral hearing loss, or noise‐induced hearing loss. The results show (1) hearing sensitivity declines more than twice as fast in men as in women at most ages and frequencies, (2) longitudinal declines in hearing sensitivity are detectable at all frequencies among men by age 30, but the age of onset of decline is later in women at most frequencies and varies by frequency in women, (3) women have more sensitive hearing than men at frequencies above 1000 Hz but men have more sensitive hearing than women at lower frequencies, (4) learning effects bias cross‐sectional and short‐term longitudinal studies, and (5) hearing levels and longitudinal patterns of change are highly variable, even in this highly selected group. These longitudinal findings document gender differences in hearing levels and show that age‐associated hearing loss occurs even in a group with relatively low‐noise occupations and with no evidence of noise‐induced hearing loss.
TL;DR: A simple forward viewing ultrasound catheter includes one or more transducers (10) and an ultrasound mirror (14) supported by a bearing (12) in a sealed end of a catheter with a drive cable imparting relative motion to the transducers and mirror as discussed by the authors.
Abstract: A simple forward viewing ultrasound catheter includes one or more transducers (10) and an ultrasound mirror (14) supported by a bearing (12) in a sealed end of a catheter (16) with a drive cable (18) imparting relative motion to the transducer (10) and mirror (14) . The mirror directs ultrasound waves forward of the catheter. An optical fiber can be provided to direct a laser beam for ablation of atheroma while under guidance of simultaneous intravascular ultrasound.
TL;DR: In this paper, a time-domain algorithm that solves the Khokhlov-Zabolotskaya-Kuznetsov (KZK) nonlinear parabolic wave equation is described.
Abstract: A time‐domain algorithm that solves the Khokhlov–Zabolotskaya–Kuznetsov (KZK) nonlinear parabolic wave equation is described. The algorithm models the propagation of pulsed finite amplitude sound beams radiated from axisymmetric sources in homogeneous, thermoviscous fluids. Numerical results are presented for waveform distortion and shock formation in directive beams radiated by pulsed circular pistons. Waveforms are calculated through the shock region and out to far‐field locations where they are dominated by the nonlinearly generated low‐frequency components. Effects of pulse duration, frequency modulation, and noise are examined. Methods for including relaxation and focusing are described.
TL;DR: A two-mass model of vocal-fold vibrations is analyzed with methods from nonlinear dynamics and it is shown that a sufficiently large tension imbalance of the left and right vocal fold induces bifurcations to subharmonic regimes, toroidal oscillations, and chaos.
Abstract: A two-mass model of vocal-fold vibrations is analyzed with methods from nonlinear dynamics. Bifurcations are located in parameter planes of physiological interest (subglottal pressure, stiffness of the folds). It is shown that a sufficiently large tension imbalance of the left and right vocal fold induces bifurcations to subharmonic regimes, toroidal oscillations, and chaos. The corresponding attractors are characterized by phase portraits, spectra, and next-maximum maps. The relevance of these simulations for voice disorders such as laryngeal paralysis is discussed.
TL;DR: Vowel tokens receiving high goodness ratings in experiment 1 were more difficult to discriminate in experiment 2 and were more tightly clustered in the MDS solutions of experiment 3, which support the existence of a perceptual magnet effect.
Abstract: Recent experiments have demonstrated that the category goodness of speech sounds strongly influences perception in both adult and infants [Kuhl, Percept. Psychophys. 50, 93-107 (1991); Kuhl et al., Science 255, 606-608 (1992)]. Stimuli judged as exceptionally good instances of phonetic categories (prototypes) make neighboring tokens in the vowel space seem more similar, exhibiting a perceptual magnet effect. Three experiments further examined the perceptual magnet effect in adults. Experiment 1 collected goodness and identification judgments for 13 variants of the vowel /i/. Experiment 2 used signal detection theory to assess the discrimination of these tokens using a bias-free measure (d'). Experiment 3 employed multidimensional scaling (MDS) to geometrically model the distortion of the perceptual space due to the magnet effect. The results demonstrated a strong relationship between category goodness and discrimination. Vowel tokens receiving high goodness ratings in experiment 1 were more difficult to discriminate in experiment 2 and were more tightly clustered in the MDS solutions of experiment 3. These findings support the existence of a perceptual magnet effect, and may help explain some aspects of first language learning in infants and second language learning in adults.
TL;DR: An angioplasty balloon catheter particularly adapted for ablation of a stenosis in vivo has a balloon which may be inflated with a conductive contrast fluid injected proximally to the balloon, and is furthermore metalized on the outside of the balloon and catheter shaft.
Abstract: An angioplasty balloon catheter particularly adapted for ablation of a stenosis in vivo has a balloon which may be inflated with a conductive contrast fluid injected proximally to the balloon, and is furthermore metalized on the outside of the balloon and catheter shaft. The balloon has piezoelectric properties, and may be excited by application of an ultrasonic signal across the balloon between the metalized surface and the contrast fluid. The catheter is guided by a centrally located guide wire to the site of the stenosis. If the distal tip of the catheter shaft cannot pass through the stenosis, excitation of the piezoelectric balloon in a deflated state at the site of the stenosis causes ultrasonic hammering vibrations at the tip of the catheter shaft which ablate the stenosis. After the tip of the catheter has hammered its way across the stenosis, and while maintaining the ultrasonic excitation signal, the balloon is inflated to keep the vibrating balloon surface in contact with the stenosis. Unlike the case with other inflatable balloon catheters which simply press the stenosis against the blood vessel wall, the stenosis is broken up by ultrasonic vibrations and is carried away by the blood flow, minimizing the risk of re-stenosis.
TL;DR: In this article, a real-time random access animation user interface environment referred to as interFACE enables a user to create and control animated lip-synchronized images or objects utilizing a personal computer for use in the users programs and products.
Abstract: A random access animation user interface environment referred to as interFACE enabling a user to create and control animated lip-synchronized images or objects utilizing a personal computer for use in the users programs and products. A real-time random-access interface driver (RAVE) together with a descriptive authoring language (RAVEL) is used to provide synthesized actors ("synactors"). The synactors may represent real or imaginary persons or animated characters, objects or scenes. The synactors may be created and programmed to perform actions including speech which are not sequentially pre-stored records of previously enacted events. Furthermore, animation and sound synchronization may be produced automatically and in real-time. Sounds and visual images of a real or imaginary person or animated character associated with those sounds are input to a system and may be decomposed into constituent parts to produce fragmentary images and sounds. A set of characteristics is utilized to define a digital model of the motions and sounds of a particular synactor. The general purpose system is provided for random access and display of synactor images on a frame-by-frame basis, which is organized and synchronized with sound. Both synthetic speech and digitized recording may provide the speech for synactors.
TL;DR: This optimized ATSP (OATSP) has an almost ideal characteristic to measure impulse responses shorter than its specific length N and it is newly shown in this paper that OATSP has also a good characteristic toMeasure impulse responses longer than N.
Abstract: Transfer functions of acoustic systems often exhibit wide dynamic ranges and very long impulse responses. A ‘‘time‐stretched’’ pulse as proposed by Aoshima (ATSP), though originally given in a very specific form seems to be one of the most promising signals to measure transfer functions with characteristics of acoustic system mentioned as above. In this paper, this pulse (ATSP) is first generalized and then optimized for the measurement of long impulse responses. This optimized ATSP (OATSP) has an almost ideal characteristic to measure impulse responses shorter than its specific length N. Moreover, it is newly shown in this paper that OATSP has also a good characteristic to measure impulse responses longer than N. Discussion is presented on how to design OATSP suitable for a specific situation of measurement by analyzing errors, when the pulse is used to measure impulse responses longer than N.
TL;DR: A method for encoding a signal that includes a speech component that is classified in one of at least two modes, based, for example, on pitch stationarity, short-term level gradient or zero crossing rate, is described.
Abstract: A method for encoding a signal that includes a speech component is described. First and second linear prediction windows of a frame are analyzed to generate sets of filter coefficients. First and second pitch analysis windows of the frame are analyzed to generate pitch estimates. The frame is classified in one of at least two modes, e.g. voiced, unvoiced and noise modes, based, for example, on pitch stationarity, short-term level gradient or zero crossing rate. Then the frame is encoded using the filter coefficients and pitch estimates in a particular manner depending upon the mode determination for the frame, preferably employing CELP based encoding algorithms.
TL;DR: In this article, the authors propose an approach for reducing acoustic background noise for use with a telephone handset (10) or a boom microphone device (100) or an audio boom headset (401) or the like.
Abstract: Apparatus for reducing acoustic background noise for use with a telephone handset (10) or a boom microphone device (100) or a boom headset (401) or the like. The apparatus includes first (12) and second (14) microphones which are arranged such that the first microphone (12) receives a desired speech input and the background noise present in the vicinity of the speech, and the second microphone (14) receives substantially only the background noise. The background noise from the second microphone (14) is converted into a corresponding electrical signal and substracted (16) from a signal corresponding to the speech and background noise obtained from the first microphone (12) so as to produce a signal representing substantially the speech.
TL;DR: In this article, a method of repairing machine-recognized speech is comprised of the steps of receiving from a recognition engine a first n-best list of hypotheses and scores for each hypothesis generated in response to a primary utterance to be recognized.
Abstract: A method of repairing machine-recognized speech is comprised of the steps of receiving from a recognition engine a first n-best list of hypotheses and scores for each hypothesis generated in response to a primary utterance to be recognized. An error within the hypothesis having the highest score is located. Control signals are generated from the first n-best list which are input to the recognition engine to constrain the generation of a second n-best list of hypotheses, and scores for each hypothesis, in response to an event independent of the primary utterance. The scores for the hypotheses in the first n-best list are combined with the scores for the hypotheses in the second n-best list. The hypothesis having the highest combined score is selected as the replacement for the located error.
TL;DR: In this paper, a speech coding system employing an adaptive codebook model of periodicity is augmented with a pitch-predictive filter (PPF), which has a delay equal to the integer component of the pitch-period and a gain which is adaptive based on a measure of the periodicity of the speech signal.
Abstract: A speech coding system employing an adaptive codebook model of periodicity is augmented with a pitch-predictive filter (PPF). This PPF has a delay equal to the integer component of the pitch-period and a gain which is adaptive based on a measure of periodicity of the speech signal. In accordance with an embodiment of the present invention, speech processing systems which include a first portion comprising an adaptive codebook and corresponding adaptive codebook amplifier and a second portion comprising a fixed codebook coupled to a pitch filter, are adapted to delay the adaptive codebook gain; determine the pitch filter gain based on the delayed adaptive codebook gain, and amplify samples of a signal in the pitch filter based on said determined pitch filter gain. The adaptive codebook gain is delayed for one subframe. The pitch filter gain equals the delayed. adaptive codebook gain, except when the adaptive codebook gain is either less than 0.2 or greater than 0.8., in which cases the pitch filter gain is set equal to 0.2 or 0.8, respectively.
TL;DR: An analytical model of the mechanical response of the ear to a sound stimulus is proposed which supports the claim that mechanical interaural coupling is the key to this animal's ability to localize sound sources.
Abstract: An analysis is presented of the mechanical response to a sound field of the ears of the parasitoid fly Ormia ochracea. This animal shows a remarkable ability to detect the direction of an incident sound stimulus even though its acoustic sensory organs are in very close proximity to each other. This close proximity causes the arrival times of the sound pressures at the two ears to be less than 1 to 2 μs depending on the direction of propagation of the sound wave. The small differences in these two pressures must be processed by the animal in order to determine the incident direction of the sound. In this fly, the ears are so close together that they are actually joined by a cuticular structure which couples their motion mechanically and subsequently magnifies interaural differences. The use of a cuticular structure as a means to couple the ears to achieve directional sensitivity is novel and has not been reported in previous studies of directional hearing. An analytical model of the mechanical response of ...
TL;DR: In this paper, the authors compare causal theories, based on Kramers-Kronig relations, fractional calculus, and on those derived from new time domain causal relationships, to diverse data.
Abstract: This study compares causal theories, based on Kramers–Kronig relations, fractional calculus, and on those derived from new time domain causal relationships, to diverse data. All these theories are based on the assumptions that the functional form of the attenuation persists beyond the measurement range and that attenuation is much smaller than the wave number. The data, for lossy media with attenuation having a power‐law frequency dependence with an exponent y, include cases for both liquids and solids, ranging from acoustic to ultrasound frequencies. Data are in closer correspondence with the new theory which predicts decreasing dispersion as the power exponent y approaches zero or an even integer. Experimental results and supporting evidence show that the classical case of frequency‐squared attenuation is dispersionless. An approximate nearly local Kramers–Kronig theory is in agreement with the time causal theory when the exponent is close to one, but deviates for other values. The comprehensive time causal theory is shown to be equivalent to two other theories derived from exact Kramers–Kronig relations and from fractional calculus and it covers the y odd integer cases which are missing or incomplete in these approaches. Attenuation–dispersion relations are presented in two forms: one for a general frequency range and another for a finite range. It is demonstrated that complete velocity dispersion (within a signal bandwidth) can be predicted from knowledge of the attenuation data and velocity at a single frequency including the velocity at either zero frequency (y≳1) and or at a high‐frequency limit (0
TL;DR: In this paper, the theory and design of a broadband array of sensors with a frequency invariant far-field beam pattern over an arbitrarily wide design bandwidth is presented, and the problem of designing a practical sensor array is treated as an approximation to this continuous sensor using a discrete set of filtered broadband omnidirectional array elements.
Abstract: The theory and design of a broadband array of sensors with a frequency invariant far‐field beam pattern over an arbitrarily wide design bandwidth is presented. The frequency invariant beam pattern property is defined in terms of a continuously distributed sensor, and the problem of designing a practical sensor array is then treated as an approximation to this continuous sensor using a discrete set of filtered broadband omnidirectional array elements. The design methodology is suitable for one‐, two‐, and three‐dimensional sensor arrays; it imposes no restrictions on the desired aperture distribution (beam shape), and can cope with arbitrarily wide bandwidths. An important consequence of the results is that the frequency response of the filter applied to the output of each sensor can be factored into two components: One component is related to a slice of the desired aperture distribution, and the other is sensor independent. The results also indicate that the locations of the sensors are not a crucial design consideration, although it is shown that nonuniform spacings simultaneously avoid spatial aliasing and minimize the number of sensors. An example design which covers a 10:1 frequency range (which is suitable for speech acquisition using a microphone array) illustrates the utility of the method. Finally, the theory is generalized to cover a parameterized class of arrays in which the frequency dependence of the beam pattern can be controlled in a continuous manner from a classical single‐frequency design to a frequency invariant design.
Abstract: In a high-energy ultrasound therapy method and apparatus, said apparatus comprises a therapy device with at least one ultrasound therapy transducer element and a signal generator supplying an electronic signal to said ultrasound transducer element, the signal generator supplying the transducer(s) with a wideband electronic signal of the random or pseudo-random type.
TL;DR: In this paper, the headset comprises left and right elongate ear pieces having respective first ends to receive ear tips and respective interconnected second ends, each ear piece includes a one-piece elongate body made of flexible and resilient material and curved outwardly to skirt round the user's head when the ear tips are applied to a user's ears.
Abstract: The headset comprises left and right elongate ear pieces having respective first ends to receive ear tips and respective, interconnected second ends. Each ear piece includes a one-piece elongate body made of flexible and resilient material and curved outwardly to skirt round the user's head when the ear tips are applied to a user's ears. The elongate body is formed with a first length of smaller cross section and therefore of greater flexibility proximate the second end of the corresponding ear piece and with a second length of lower flexibility situated between that first length and the corresponding ear tip. Bending of the elongate bodies when the ear pieces are spread apart is therefore concentrated in the first lengths and pressure applied to the user's ears by the ear tips is mainly produced by the bent first lengths and transmitted to the ear tips through the second lengths of lower flexibility. The elongate ear pieces have respective second end sections assembled laterally adjacent to each other, and a threaded sleeve is mounted on these laterally adjacent second end sections and is rotated and displaced longitudinally to adjust the level of pressure applied to the user's ears by the ear tips.
TL;DR: In this paper, a hearing compensation system for the hearing impaired comprises a plurality of bandpass filters having an input connected to an input transducer and each bandpass filter having an output connected to the input of one of a multiplicative AGC circuits.
Abstract: A hearing compensation system for the hearing impaired comprises a plurality of bandpass filters having an input connected to an input transducer and each bandpass filter having an output connected to the input of one of a plurality of multiplicative AGC circuits whose outputs are summed together and connected to the input of an output transducer. The multiplicative AGC circuits attenuate acoustic signals having a constant background level without the loss of speech intelligibility. The identification of the background noise portion of the acoustic signal is made by the constancy of the envelope of the input signal in each of the several frequency bands. The background noise that will be suppressed includes multi-talker speech babble, fan noise, feedback whistle, fluorescent light hum, and white noise.