scispace - formally typeset
Search or ask a question

Showing papers in "Journal of the Acoustical Society of America in 1986"


PatentDOI
TL;DR: In this paper, a method and apparatus for periodically interrupting ultrasonic power applied to a vibrating tip to control its amplitude between high and low or zero amplitudes with a selectible duty cycle and repetition rate provides enhanced fragmentation and improves surgical control.
Abstract: A method and apparatus for periodically interrupting ultrasonic power applied to a ultrasonically vibrating tip to control its amplitude between high and low or zero amplitudes with a selectible duty cycle and repetition rate provides enhanced fragmentation and improves surgical control. The duty cycle may also vary as a function of a remotely sensed parameter such as tissue temperature.

897 citations


PatentDOI
TL;DR: In this paper, an ultrasonic transducer is carried by the distal end of a catheter adapted for insertion into a vessel, and either the transducers or another element is rotated and/or translated relative to the catheter to image different portions of the vessel.
Abstract: Ultrasonic apparatus, system and method for high resolution intravascular imaging to assist indovascular lesions and to monitor the results of interventional therapy. An ultrasonic transducer is carried by the distal end of a catheter adapted for insertion into a vessel, and either the transducer or another element is rotated and/or translated relative to the catheter to image different portions of the vessel.

475 citations


Journal ArticleDOI
TL;DR: These findings agree with independently observed perceptual results and support Stevens' quantal theory of vowel production and perceptual constraints on production predicted from the critical bark difference level of the model.
Abstract: A quantitative perceptual model of human vowel recognition based upon psychoacoustic and speech perception data is described. At an intermediate auditory stage of processing, the specific bark difference level of the model represents the pattern of peripheral auditory excitation as the distance in critical bands (barks) between neighboring formants and between the fundamental frequency (F0) and first formant (F1). At a higher, phonetic stage of processing, represented by the critical bark difference level of the model, the transformed vowels may be dichotomously classified based on whether the difference between formants in each dimension falls within or exceeds the critical distance of 3 bark for the spectral center of gravity effect [Chistovich et al., Hear. Res. 1, 185–195 (1979)]. Vowel transformations and classifications correspond well to several major phonetic dimensions and features by which vowels are perceived and traditionally classified. The F1–F0 dimension represents vowel height, and high vo...

452 citations


Journal ArticleDOI
TL;DR: The shape of the auditory filter was estimated at three center frequencies, 0.5, 1.0, and 2.0 kHz, for five subjects with unilateral cochlear impairments and six subjects with bilateral impairments, finding that the equivalent rectangular bandwidth of the filter was always greater for the impaired ear than for the normal ear at each center frequency.
Abstract: The shape of the auditory filter was estimated at three center frequencies, 0.5, 1.0, and 2.0 kHz, for five subjects with unilateral cochlear impairments. Additional measurements were made at 1.0 kHz using one subject with a unilateral impairment and six subjects with bilateral impairments. Subjects were chosen who had thresholds in the impaired ears which were relatively flat as a function of frequency and ranged from 15 to 70 dB HL. The filter shapes were estimated by measuring thresholds for sinusoidal signals (frequency f) in the presence of two bands of noise, 0.4 f wide, one above and one below f. The spectrum level of the noise was 50 dB (re: 20 mu Pa) and the noise bands were placed both symmetrically and asymmetrically about the signal frequency. The deviation of the nearer edge of each noise band from f varied from 0.0 to 0.8 f. For the normal ears, the filters were markedly asymmetric for center frequencies of 1.0 and 2.0 kHz, the high-frequency branch being steeper. At 0.5 kHz, the filters were more symmetric. For the impaired ears, the filter shapes varied considerably from one subject to another. For most subjects, the lower branch of the filter was much less steep than normal. The upper branch was often less steep than normal, but a few subjects showed a near normal upper branch. For the subjects with unilateral impairments, the equivalent rectangular bandwidth of the filter was always greater for the impaired ear than for the normal ear at each center frequency. For three subjects at 0.5 kHz and one subject at 1.0 kHz, the filter had too little selectivity for its shape to be determined.

418 citations


Journal ArticleDOI
TL;DR: A linear, mathematical model of cochlear biomechanics is presented and active elements are intended to represent the motile action of outer hair cells to simulating the high sensitivity and sharp tuning characteristic of the mammalian cochlea.
Abstract: A linear, mathematical model of cochlear biomechanics is presented in this paper. In this model, active elements are essential for simulating the high sensitivity and sharp tuning characteristic of the mammalian cochlea. The active elements are intended to represent the motile action of outer hair cells; they are postulated to be mechanical force generators that are powered by electrochemical energy of the cochlear endolymph, controlled by the bending of outer hair cell stereocilia, and bidirectionally coupled to cochlear partition mechanics. The active elements are spatially distributed and function collectively as a cochlear amplifier. Excessive gain in the cochlear amplifier causes spontaneous oscillations and thereby generates spontaneous otoacoustic emissions.

377 citations


Journal ArticleDOI
TL;DR: High-frequency plateaus were observed in both isovelocity tuning curves and phase-frequency curves, and there is a progressive loss of sensitivity of the mechanical response with time for the frequencies around CF, but not for frequencies on the tail of the tuning curve.
Abstract: Basilar membrane (BM) velocity was measured at a site 3.5 mm from the basal end of the chinchilla cochlea using the Mossbauer technique. The threshold of the compound action potential recorded at the round window in response to tone bursts was used as an indicator of the physiological state of the cochlea. The BM input-output functions display a compressive nonlinearity for frequencies around the characteristic frequency (CF, 8 to 8.75 kHz), but are linear for frequencies below 7 and above 10.5 kHz. In preparations with little surgical damage, isovelocity tuning curves at 0.1 mm/s are sharply tuned, have Q10's of about 6, minima as low as 13 dB SPL, tip-to-tail ratios (at 1 kHz) of 56 to 76 dB, and high-frequency slopes of about 300 dB/oct. These mechanical responses are as sharply tuned as frequency-threshold curves of chinchilla auditory nerve fibers with corresponding CF. There is a progressive loss of sensitivity of the mechanical response with time for the frequencies around CF, but not for frequencies on the tail of the tuning curve. In some experiments the nonlinearity was maintained for several hours, in spite of a considerable loss of sensitivity of the BM response. High-frequency plateaus were observed in both isovelocity tuning curves and phase-frequency curves.

349 citations


Journal ArticleDOI
TL;DR: A probabilistic model is described for transmitter release from hair cells, auditory neuron EPSP's, and discharge patterns that mimics successfully the adaptation of successive EPSP amplitudes of the afferent neuron of the goldfish sacculus and offers a reinterpretation of the implications of these studies for hair cell synaptic mechanism.
Abstract: A probabilistic model is described for transmitter release from hair cells, auditory neuron EPSP’s, and discharge patterns. The model assumes that the release fraction of the transmitter is a function of stimulus intensity. It further assumes that some of this transmitter substance is taken back into the cell while some is irretrievably lost from the cleft. These assumptions differ from other recent models which propose multiple release sites, fixed release fractions, and no transmitter reuptake. The model produces realistic mammalian rate intensity functions, interval and period histograms, incremental responses, and adaptation effects. It mimics successfully the adaptation of successive EPSP amplitudes of the afferent neuron of the goldfish sacculus and offers a reinterpretation of the implications of these studies for hair cell synaptic mechanism.

345 citations


Journal ArticleDOI
TL;DR: In this paper, the composite roughness model is applied to bottom backscattering in the frequency range 10-100 kHz and the Kirchhoff approximation is used to obtain better results.
Abstract: The composite roughness model is applied to bottom backscattering in the frequency range 10–100 kHz. For angles near normal incidence, the composite roughness model is replaced by the Kirchhoff approximation which gives better results. In addition, sediment volume scattering is treated, with account taken of refraction and reflection at the randomly sloping interface. In applying the model to published data it is found that sediment volume scattering is dominant in soft sediments except at small and large grazing angles. For coarse sand bottoms, roughness scattering dominates over a wide range of grazing angles. Implications for acoustic remote sensing are discussed.

343 citations


PatentDOI
TL;DR: In this article, the authors proposed a text locating system that allows a user to switch between a dictation mode, which inserts recognized words into text, and a search mode which uses them to search for new cursor locations.
Abstract: A text locating system recognizes spoken utterances, uses the recognized words as a search string, and searches text for words matching that search string. The probability that a given vocabulary word is selected as a search word is altered both by limiting the recognizable vocabulary to words in the text to the searched, and by altering the probability that individual recognizable words will be selected as a function of the number of time they occur in that text. The system performs incremental searches by adding successively recognized words to the search string and searching for the next occurrence of the string in response to each such addition. The invention can be used in a text editing system which enables a user to switch between a dictation mode, which inserts recognized words into text, and a search mode, which uses them to search for new cursor locations. Broadly speaking, the invention provides a computer system which recognizes spoken words, which has a data structure representing words; which uses that data structure for a purpose other than speech recognition; and which alters the probability that a given vocabulary word will be recognized as a function of the frequency of that word in the data structure.

314 citations


Journal ArticleDOI
TL;DR: The use of an auxiliary random noise generator for this modeling is described, which is easy to implement, provides continuous on‐line modeling, and has minimal effect on the final value of the error signal.
Abstract: Active sound attenuation systems may be described using a system identification framework in which an adaptive filter is used to model the performance of an unknown acoustical plant. An error signal may be obtained from a location following an acoustical summing junction where the undesired noise is combined with the output of a secondary sound source. In order for the model output to properly converge to a value that will minimize the error signal, it is frequently necessary to determine the transfer function of the secondary sound source and the path to the error signal measurement. Since these transfer functions are continuously changing in a real system, it is desirable to perform continuous on‐line modeling of the output transducer and error path. In this paper, the use of an auxiliary random noise generator for this modeling is described. Based on a Galois sequence, this technique is easy to implement, provides continuous on‐line modeling, and has minimal effect on the final value of the error signal.

304 citations


Journal ArticleDOI
TL;DR: This paper performed statistical analysis of F1 and F2 measurements from nucleus and offglide sections of isolated Canadian English vowels and found significant formant frequency change not only for the phonetic diphthongs, but also for the monophthongs.
Abstract: Statistical analysis of F1 and F2 measurements from nucleus and offglide sections of isolated Canadian English vowels shows significant formant frequency change not only for the ‘‘phonetic diphthongs’’ /e/ and /o/, but also for the ‘‘monophthongs’’ /ι/, /q/, and /1/. In a perceptual experiment, brief sections were extracted from ‘‘nucleus’’ and ‘‘offglide’’ portions of naturally produced vowels. Two sections from each vowel were presented to listeners in each of three conditions: (1) natural order (nucleus followed by offglide); (2) repeated nucleus (nucleus followed by itself); and (3) reverse (offglide followed by nucleus). Listeners’ error rates for the natural order condition were comparable to those for unmodified full vowels (averaging 14% and 13%, respectively). Significantly higher error rates were found for the repeated nucleus (32%) and reverse (38%) conditions. Observed confusion matrices were strongly correlated with predictions from a pattern recognition model incorporating the formant measur...

Journal ArticleDOI
TL;DR: In this paper, a systematic investigation of the various measurement errors that can occur and their effect on the calculated quantities is made, and conclusions concerning the most favorable measurement configuration to avoid these errors are drawn.
Abstract: Using the two‐microphone method, acoustic properties in ducts, as, for example, reflection coefficient and acoustic impedance, can be calculated from a transfer function measurement between two microphones. In this paper, a systematic investigation of the various measurement errors that can occur and their effect on the calculated quantities is made. The input data for the calculations are the measured transfer function, the microphone separation, and the distance between one microphone and the sample. First, errors in the estimate of the transfer function are treated. Conclusions concerning the most favorable measurement configuration to avoid these errors are drawn. Next, the length measurement errors are treated. Measurements were made to study the question of microphone interference. The influence of errors on the calculated quantities has been investigated by numerical simulation. From this, conclusions are drawn on the useful frequency range for a given microphone separation and on the magnitude of errors to expect for different cases.

Journal ArticleDOI
TL;DR: A model of interaural cross correlation is extended by a "contralateral-inhibition mechanism" and by "monaural detectors" in order to simulate a wide range of psychoacoustic lateralization data.
Abstract: Running interaural cross correlation is a basic assumption to model the performance of the binaural auditory system. Although this concept is particularly suited to simulate psychoacoustic localization phenomena, there exist some localization effects which cannot be explained by pure cross correlation. In this paper a model of interaural cross correlation is extended by a ‘‘contralateral‐inhibition mechanism’’ and by ‘‘monaural detectors’’ in order to simulate a wide range of psychoacoustic lateralization data. The extended model explains lateralization of pure tones with interaural time differences as well as with interaural level differences. Multiple images are predicted for tones with characteristic combinations of interaural signal parameters and for noise signals with different degrees of interaural cross correlation. The model is also capable of simulating dynamic lateralization phenomena, such as the ‘‘law of the first wave front’’ which is dealt with in a companion paper [Lindemann, J. Acoust. So...

PatentDOI
TL;DR: In this article, an annular array ultrasound transducer, a mechanical driver, a transmitter/receiver, and a controller are used to change at least one of the position and direction of ultrasound beams transmitted and received by the transducers.
Abstract: An ultrasound therapy system comprises an annular array ultrasound transducer, a mechanical driver, a transmitter/receiver, and a controller. The controller includes an imaging controller, a heating controller, and a select controller. The mechanical driver mechanically drives the transducer to change at least one of the position and direction of ultrasound beams transmitted and received by the transducer. The transmitter/receiver may supply drive signals to the respective elements of the transducer, and receive the ultrasound echo signals from the elements. The imaging controller gives a first drive command to the transmitter/receiver, and at the same time drives the mechanical driver, radiates scanning ultrasound beams for tomographing through the transducer, and obtains a tomogram of a target portion in a patient from the echo signal derived from the transmitter/receiver. The heating controller drives the transmitter/receiver by a second drive signal to cause the transducer to radiate heating ultrasound beams. The heating ultrasound beams heat the target. These imaging and heating controllers are selectively activated by the select controller.

Journal ArticleDOI
TL;DR: The relationship between dynamic spectral features and the identification of Japanese syllables modified by initial and/or final truncation is examined, suggesting that crucial information for both vowel and consonant identification is contained across the same initial part of each syllable.
Abstract: This paper examines the relationship between dynamic spectral features and the identification of Japanese syllables modified by initial and/or final truncation. The experiments confirm several main points. ‘‘Perceptual critical points,’’ where the percent correct identification of the truncated syllable as a function of the truncation position changes abruptly, are related to maximum spectral transition positions. A speech wave of approximately 10 ms in duration that includes the maximum spectral transition position bears the most important information for consonant and syllable perception. Consonant and vowel identification scores simultaneously change as a function of the truncation position in the short period, including the 10‐ms period for final truncation. This suggests that crucial information for both vowel and consonant identification is contained across the same initial part of each syllable. The spectral transition is more crucial than unvoiced and buzz bar periods for consonant (syllable) perception, although the latter features are of some perceptual importance. Also, vowel nuclei are not necessary for either vowel or syllable perception.

Journal ArticleDOI
TL;DR: Experiments with the voice samples show that the NNE is especially effective for detecting glottic cancer, recurrent nerve paralysis, and vocal cord nodules.
Abstract: In order to evaluate noise components included in pathologic voice signals, a novel acoustic measure, normalized noise energy (NNE), is proposed and its effectiveness for the detection of laryngeal pathologies is investigated with 250 vowel samples spoken by 64 control (normal) subjects and 186 patients with various laryngeal diseases. The NNE is automatically computed from the voice signals using an adaptive comb filtering method performed in the frequency domain. Experiments with the voice samples show that the NNE is especially effective for detecting glottic cancer, recurrent nerve paralysis, and vocal cord nodules. Specifically, when glottic cancer is represented in terms of the T classification adopted by the UICC (Union Internationale Contre le Cancer), glottic T2-T4 cancer can be perfectly discriminated from normal samples, but 22.6% of patients with glottic T1 cancer are incorrectly classified as normal, with an error rate of 9.4% for normal subjects.

Journal ArticleDOI
TL;DR: Computer-aided acoustical analysis of word durations showed a localized, large magnitude increase in the duration of the focused word for both statements and questions, and analysis of F0 revealed a more complex pattern of results, with the shape of the F0 topline dependent on sentence type and focus location.
Abstract: An acoustical study of speech production was conducted to determine the manner in which the location of linguistic focus influences intonational attributes of duration and fundamental voice frequency (F0) in matched statements and questions. Speakers orally read sentences that were preceded by aurally presented stimuli designed to elicit either no focus or focus on the first or last noun phrase of the target sentences. Computer‐aided acoustical analysis of word durations showed a localized, large magnitude increase in the duration of the focused word for both statements and questions. Analysis of F0 revealed a more complex pattern of results, with the shape of the F0 topline dependent on sentence type and focus location. For sentences with neutral or sentence‐final focus, the difference in the F0 topline between questions and statements was evident only on the last key word, where the F0 peak of questions was considerably higher than that of statements. For sentences with focus on the first key word, ther...

Journal ArticleDOI
TL;DR: In this paper, closed-form asymptotic expressions for the frequency-wavenumber dispersion relations in doubly rotated quartz plates vibrating in the vicinity of the odd pure thickness frequencies are derived from the equations of linear piezoelectricity and the associated boundary conditions on the major surfaces.
Abstract: Closed‐form asymptotic expressions for the frequency–wavenumber dispersion relations in doubly rotated quartz plates vibrating in the vicinity of the odd pure thickness frequencies are derived from the equations of linear piezoelectricity and the associated boundary conditions on the major surfaces. The usual assumptions of small piezoelectric coupling and small wavenumbers along the plate are made and it is supposed that the pure thickness frequencies are sufficiently different that one pure thickness wave is dominant at a time. In the treatment the mechanical displacement is decomposed along the eigenvector triad of the pure thickness solution to facilitate the asymptotic analysis. The fact that the wavenumbers along the plate are restricted to be small significantly reduces the complexity of the equations without neglecting any transformed elastic constants. The resulting asymptotic dispersion equation enables the construction of a scalar differential equation describing the transverse behavior of essentially thickness modes of vibration in doubly rotated quartz plates. The scalar equation is applied in the analysis of both trapped energy resonators with rectangular electrodes and contoured crystal resonators using established procedures. In particular, calculations performed for the contoured SC cut and a number of other doubly rotated orientations are shown to be in excellent agreement with experiment. Since the differential equation for each harmonic family depends on the order of the harmonic and in the general doubly rotated case contains mixed derivatives in the plane of the plate, a different transformation is required for each harmonic family to obtain the coordinate system in which the mixed derivatives do not appear and, hence, the equation is separable. An interesting consequence of this transformation is that since the nodal planes of the anharmonics of each harmonic family of the contoured SC‐cut quartz resonator are oriented along the transformed coordinate system for that harmonic family, they are oriented differently for each harmonic family.

PatentDOI
TL;DR: In this article, a noninvasive system and method for inducing vibrations in a selected element of the human body and detecting the nature of responses for determining mechanical characteristics of the element are provided.
Abstract: A non-invasive system and method for inducing vibrations in a selected element of the human body and detecting the nature of responses for determining mechanical characteristics of the element are provided. The method comprises the steps of: inducing multiple-frequency vibrations, including below 20 KHz, in a selected element of the body by use of a driver; determining parameters of the vibration exerted on the body by the driver; sensing variations of a dimension of the element of the body over time, including in response to the driver; correlating the variations with frequency components of operation of the driver below 20 KHz to determine corresponding frequency components of the variations; resolving the frequency components into components of vibration mode shape; and determining the mechanical characteristics of the element on the basis of the parameters of vibration exerted by the driver and of the components of vibration mode shape.

Journal ArticleDOI
TL;DR: This experiment estimated the degree of mistuning required for this phenomenon to occur, for complex tones with 10 or 12 equal-amplitude components (60 dB SPL per component), in terms of a hypothetical harmonic sieve and mechanisms for the formation of perceptual streams.
Abstract: When a low harmonic in a harmonic complex tone is mistuned from its harmonic value by a sufficient amount it is heard as a separate tone, standing out from the complex as a whole. This experiment estimated the degree of mistuning required for this phenomenon to occur, for complex tones with 10 or 12 equal‐amplitude components (60 dB SPL per component). On each trial the subject was presented with a complex tone which either had all its partials at harmonic frequencies or had one partial mistuned from its harmonic frequency. The subject had to indicate whether he heard a single complex tone with one pitch or a complex tone plus a pure tone which did not ‘‘belong’’ to the complex. An adaptive procedure was used to track the degree of mistuning required to achieve a d’ value of 1. Threshold was determined for each ot the first six harmonics of each complex tone. In one set of conditions stimulus duration was held constant at 410 ms, and the fundamental frequency was either 100, 200, or 400 Hz. For most conditions the thresholds fell between 1% and 3% of the harmonic frequency, depending on the subject. However, thresholds tended to be greater for the first two harmonics of the 100‐Hz fundamental and, for some subjects, thresholds increased for the fifth and sixth harmonics. In a second set of conditions fundamental frequency was held constant at 200 Hz, and the duration was either 50, 110, 410, or 1610 ms. Thresholds increased by a factor of 3–5 as duration was decreased from 1610 ms to 50 ms. The results are discussed in terms of a hypothetical harmonic sieve and mechanisms for the formation of perceptual streams.

Journal ArticleDOI
TL;DR: The recent report of the National Science Board's Commission on Pre-College Education in Mathematics, Science, and Technology represents a radical departure from the movements of the 50s and 60s as discussed by the authors.
Abstract: Recent national commission reports have discussed a new era for science education. Unlike the sciences of the 1950s and 1960s, this new era has raised some provocative questions concerning science education for the future. The science movements of the post‐sputnik period stressed science for the elite, more science and mathematics for the college‐bound student, developing a corps of the very best engineers and scientists who could take us to the moon and let us assume the leadership in the international scientific enterprise. It was a time whose purpose was to get the most able students from our high schools into the calculus and physics courses at universities and colleges as quickly as possible. The recent report of the National Science Board's Commission on Precollege Education in Mathematics, Science, and Technology represents a radical departure from the movements of the 50s and 60s. The purpose of this paper is to discuss the implications of their recommendations for the future of science education.

PatentDOI
TL;DR: In this article, a digital text-to-speech conversion system of the type usually contained in all-software form on a floppy disk is presented, which provides compression techniques and anti-distortion techniques which interact to provide clear speech at widely varying speeds with a minimum of memory.
Abstract: In a digital text-to-speech conversion system of the type usually contained in all-software form on a floppy disk, memory requirements are reduced while speech quality is improved, by providing compression techniques and anti-distortion techniques which interact to provide clear speech at widely varying speeds with a minimum of memory. These techniques include using Huffman coding to advantage by encoding only differences between successive waveforms where feasible, relocating delta tables and using them repetitively, using a demi-diphone organization of the speech to allow use of the same instruction lists for several sounds; and combining selective deletion or repetition of waveforms with selective interpolation to vary speed without slurring.

Journal ArticleDOI
TL;DR: Using this "low-pass impulse" method, reverberant rooms can be simulated with sufficient accuracy to investigate multiple-microphone systems that are sensitive to interchannel phase.
Abstract: A method is presented for simulating the impulse response between an acoustic source and multiple microphones in a reverberant room. The method is similar to the image method described by Allen and Berkley [J. Acoust. Soc. Am. 65, 943–950 (1979)] but includes modifications to simulate received echo arrival time accurately. The essential modification is to represent each received echo as a low‐pass‐filtered impulse at the correct arrival time. Using this ‘‘low‐pass impulse’’ method, reverberant rooms can be simulated with sufficient accuracy to investigate multiple‐microphone systems that are sensitive to interchannel phase.

Journal ArticleDOI
TL;DR: There appears to be sufficient information in the rate response of a small number of auditory nerve fibers to support behaviorally observed levels of detection performance, especially at high noise levels.
Abstract: The rate responses of auditory nerve fibers were measured for best frequency (BF) tone bursts in the presence of continuous background noise. Rate functions for BF tones were constructed over a 32‐dB range of levels, centered on the behavioral masked thresholds of cats. The tone level at which noticeable rate changes are evoked by the tones corresponds closely to behavioral masked threshold at all noise levels used (−10‐ to 30‐dB spectrum level). As the noise level increases, the response rate to the background noise approaches saturation, and the incremental rate response to tones decreases. At high noise levels, the rate responses to tones of low and medium spontaneous rate fibers are larger than those of high spontaneous rate fibers. Empirical statistics of auditory nerve fiber spike counts are reported; these differ from those expected of a Poisson process in that the variance is smaller than the mean. A new measure of discharge rate is described that allows rate changes to be expressed in units of a standard deviation. This measure allows tone‐evoked responses to be interpreted in terms of their detectability in a signal detection task. Rate responses of low and medium spontaneous rate fibers are more detectable than those of high spontaneous rate fibers, especially at high noise levels. There appears to be sufficient information in the rate response of a small number of auditory nerve fibers to support behaviorally observed levels of detection performance.

PatentDOI
TL;DR: In this paper, a decisional circuitry monitors the microphone signal of the associated microphone with respect to a MAX bus which carries microphone signals representative of the level of microphone signals at the other microphones.
Abstract: A microphone and loudspeaker arrangement for use in a teleconference system, wherein a plurality of microphones are held in a fixed relationship to a loudspeaker. The microphones are independently gated ON in response to (1) speech picked up by the microphone, (2) a loudspeaker signal driving the loudspeaker and (3) an electrical signal related to the microphone signals of the other associated microphones. A noise adapting threshold circuit generates a voltage level representative of background noise which is compared with the microphone signal of a respective microphone for determining whether the microphone is receiving speech. A decisional circuitry monitors the microphone signal of the associated microphone with respect to a MAX bus which carries microphone signals representative of the level of microphone signals at the other microphones. The decisional circuitry generates a signal indicating that the associated microphone is the first loudest microphone signal.

PatentDOI
TL;DR: In this article, a method and apparatus for measuring fluid characteristics using a non-invasive ultrasonic system for generating spatially extended signals into a volume of a fluid and detecting said signals for measuring the characteristic of the fluid.
Abstract: A method and apparatus for measuring fluid characteristics use a non-invasive ultrasonic system for generating spatially extended signals into a volume of a fluid and detecting said signals for measuring the characteristic of the fluid. The generated waves are Rayleigh-like surface waves creating, in effect, an extended aperture transducer (10) from which the waves leak into the fluid. The Rayleigh-like surface wave operates in an environment wherein the plate or other structure (14) on which the surface wave is generated has a thickness, normal to the primary direction of propagation of the wave, of less than four Rayleigh wavelengths and greater than approximately one-half of a Rayleigh wavelength. The extended aperture has a length of at least about ten Rayleigh wavelengths. The excitation for the system is generally a short pulse interrogation in order to avoid those interferences which may cause Lamb waves to be set up in the solid material. Several different configurations employing the Rayleigh-like surface wave are illustrated and discussed.

Journal ArticleDOI
TL;DR: In this paper, a speech intelligibility test and acoustical measurements were made in ten occupied classrooms and the interrelationships of these measures were considered to evaluate which were most appropriate in classrooms.
Abstract: Speech intelligibility tests and acoustical measurements were made in ten occupied classrooms. Octave‐band measurements of background noise levels, early decay times, and reverberation times, as well as various early/late sound ratios, and the center time were obtained. Various octave‐band useful/detrimental ratios were calculated along with the speech transmission index. The interrelationships of these measures were considered to evaluate which were most appropriate in classrooms, and the best predictors of speech intelligibility scores were identified. From these results ideal design goals for acoustical conditions for classrooms were determined either in terms of the 50‐ms useful/detrimental ratios or from combinations of the reverberation time and background noise level.

Journal ArticleDOI
TL;DR: Three experiments investigated subjects' ability to detect and discriminate the simulated horizontal motion of auditory targets in an anechoic environment, suggesting that for the range of simulated velocities employed in these experiments, subjects respond to spatial changes--not velocity per se--when presented with a " Motion detection or discrimination task.
Abstract: Three experiments investigated subjects’ ability to detect and discriminate the simulated horizontal motion of auditory targets in an anechoic environment. ‘‘Moving’’ stimuli were produced by dynamic application of stereophonic balancing algorithms to a two‐loudspeaker system with a 30° separation. All stimuli were 500‐Hz tones. In experiment 1, subjects had to discriminate a left‐to‐right moving stimulus from a stationary stimulus pulsed for the same duration (300 or 600 ms). For both durations, minimum audible ‘‘movement’’ angles (‘‘MAMA’s’’) were on the order of 5° for stimuli presented at 0° azimuth (straight ahead), and increased to greater than 30° for stimuli presented at ±90° azimuth. Experiment 2 further investigated MAMA’s at 0° azimuth, employing two different procedures to track threshold: holding stimulus duration constant (at 100–600 ms) while varying velocity; or holding the velocity constant (at 22°–360°/s) while varying duration. Results from the two procedures agreed with each other and ...

PatentDOI
TL;DR: In this paper, a first speech recognition method receives an acoustic description of an utterance to be recognized and scores a portion of that description against each of a plurality of cluster models representing similar sounds from different words.
Abstract: A first speech recognition method receives an acoustic description of an utterance to be recognized and scores a portion of that description against each of a plurality of cluster models representing similar sounds from different words. The resulting score for each cluster is used to calculate a word score for each word represented by that cluster. Preferably these word scores are used to prefilter vocabulary words, and the description of the utterance includes a succession of acoustic decriptions which are compared by linear time alignment against a succession of acoustic models. A second speech recognition method is also provided which matches an acoustic model with each of a succession of acoustic descriptions of an utterance to be recognized. Each of these models has a probability score for each vocabulary word. The probability scores for each word associated with the matching acoustic models are combined to form a total score for that word. The preferred speech recognition method calculates to separate word scores for each currently active vocabulary word from a common succession of sounds. Preferably the first scores is calculated by a time alignment method, while the second score is calculated by a time independent method. Preferably this calculation of two separate word scores is used in one of multiple word-selecting phase of a recognition process, such as in the prefiltering phase.

PatentDOI
TL;DR: In this paper, the LPC (linear preductive coding) filter was used to reduce the error between the input and regenerated speech signals, and the selection process involved derivation of an initial estimate followed by an iterative adjustment process in which pulses having a low energy contribution were tested in alternative positions and transferred to them if a reduced error results.
Abstract: Speech is coded such that it can be generated by a pulse excitation sequence filtered by an LPC (linear preductive coding) filter. The sequence contains, in each of successive frame periods, pulses whose positions and amplitudes may be varied. These variables are selected at the coding end to reduce the error between the input and regenerated speech signals. The selection process involves derivation of an initial estimate followed by an iterative adjustment process in which pulses having a low energy contribution are tested in alternative positions and transferred to them if a reduced error results.