Robert Berkovitz
Dolby Laboratories
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Featured researches published by Robert Berkovitz.
Journal of the Acoustical Society of America | 1992
Kenneth N. Stevens; David M. Green; Robert Berkovitz
An apparatus for and method of applying a selected range of high frequency acoustic energy to the apical end of the eardrum of an ear canal at a predetermined, substantially absolute sound pressure level at any frequency in the range of interest. The selected range of frequencies is generated at a location remote from the ear canal and the sound pressure level of each frequency is varied at the time of generation in accordance with a calibration function to ensure the frequencies reaching the apical end of the eardrum are at the predetermined sound pressure level. The calibration function is calculated by transmitting an acoustic pulse of broad frequency spectrum into the ear canal, measuring the sound pressure of the transmitted acoustic pulse and reflection thereof adjacent the entrance of the ear canal, and removing from the spectrum of the measured sound pressure the destructive interference effects between the transmitted and reflected acoustic pulses. The resultant sound pressure level contains nonuniformities having information regarding the sound transmission characteristics of the ear canal and the acoustic pulse generation and transmission system from which the calibration function is calculated.
Journal of the Acoustical Society of America | 1987
Kenneth N. Stevens; Robert Berkovitz; Gerald Kidd; David M. Green
A procedure is described for determining the absolute sound pressure at the inner end of the ear canal when a sound source is coupled to the ear, for frequencies in the range 8-20 kHz. The transducer that generates the sound is coupled to the ear canal through a lossy tube, yielding a source impedance that is approximately matched to the characteristic impedance of the ear canal. A small microphone is located in the coupling tube close to the entrance to the ear canal. Calibration is carried out by measuring the response at this microphone when an impulse is applied at the transducer. To estimate the sound pressure at the medial end of the ear canal, the Fourier transform of this impulse response is corrected by an all-pole function in which the poles are estimated from the minima in this Fourier transform. Data on individual ear canals are presented in terms of gain functions relating the sound pressure at the medial end of the ear canal to the sound pressure when the coupling tube is blocked. The average gain function for a group of adult ears increases from 2 to 12 dB over the frequency range 8-20 kHz, in rough agreement with data from ear-canal models. Possible sources of error in the calibration procedure are discussed.
Journal of the Acoustical Society of America | 1979
Robert Berkovitz
A sound reproduction arrangement and audio signal delay apparatus suitable for use in such an arrangement, includes, in a listening area, one or more primary sound sources for mono- or multichannel reproduction respectively, and a plurality of secondary sources spaced around the perimeter of the listening area and each providing an acoustic signal corresponding to that from the primary source or sources delayed by a respective and independently selectable delay constant and attenuated by a respective and independently selectable gain constant. Control apparatus is connected with the sound sources to derive a plurality of signals for the respective sound sources from an input signal. The delay constants are substantially greater than the time taken for sound to traverse the listening area, so that the sound from the secondary sources simulates reverberation or reflected sound, and a listener has a sensation of spaciousness normally associated with an enclosure which is larger than the listening area, such as a concert hall, for example. Features of the invention include a random-access memory which stores time-multiplexed data representing sampled input audio and channel-control information.
Journal of the Acoustical Society of America | 1979
Robert Berkovitz; Kenneth James Gundry
A network is described which processes two channel stereophonic sound programs for playback through left, right and center sound reproduction channels. Use of the network allows certain advantages of a three channel program to be obtained from the two channel program. The network delays the left and right channels to give precedence to the center channel information but reduces the level of this information to enable the left and right channels to establish a stereo image.
Journal of the Acoustical Society of America | 1979
James M. Kates; Robert Berkovitz
Variations in the amplitude-frequency response characteristic of a loudspeaker are compensated by a drive circuit in which an audio input signal is summed with a plurality of delayed replicas of the input signal to provide a drive signal for the loudspeaker. The delayed replicas are derived by discrete time filtering. Noise reduction techniques applied to the least delayed of the plurality of delay signals substantially improves the S/N ratio in the drive signal.
Journal of the Acoustical Society of America | 1985
Gerald Kidd; Robert Berkovitz; Kenneth N. Stevens; David M. Green
The hearing thresholds for 40 young adults (18–26 years) were measured at 13 frequencies (8, 9,…,20 kHz) using a newly developed high‐frequency audiometer. All subjects were screened at 15‐dB hearing level at the low audiometric frequencies, had tympanometry within normal limits, and had no history of significant hearing problems. The audiometer delivered sound to the ear canal through a tube and earpiece in which a small calibrated microphone was located. For each ear, a calibration function relating sound pressure at the inner end of the ear canal to voltage at the acoustic source was calculated from an impulse recorded at the microphone. Calibration functions for different ears extended over a range of 15 dB at some frequencies. For some subjects cross modes in the ear canal prevented accurate calibration at the highest frequencies. Most high‐frequency audiograms exhibited a sharply sloping increase in threshold. The location in frequency of this abrupt increase in threshold differed considerably among...
Journal of the Acoustical Society of America | 1992
J. M. Pickett; Robert Berkovitz
Instructors of courses in speech communication listed in the 1990 JASA summary of acoustics courses were surveyed to develop information about numbers and types of students, course names, current text use, available computers, and rating‐scale opinions on the potential value of computer‐based courses in acoustic phonetics. Responses were overwhelmingly positive. Surprisingly large numbers of courses and students were reported. Equipment was indicated as available for computer‐based instruction. Detailed results will be presented on the distribution of courses over academic disciplines and levels of texts in use, to characterize the needs for course development. [Work supported by NIDOCD.]
Journal of the Acoustical Society of America | 1992
Robert Berkovitz; J. M. Pickett; David R. Williams
Decreasing prices of personal computers and signal‐processing boards permit the widespread use of computer‐based instruction in acoustic phonetics. Software already developed for the Sensimetrics SpeechStation provides simplified speech synthesis and convenient spectral analysis displays. Synthesis exercises in which the student controls a vocal tract diagram are being developed to teach the principles of relations between articulatory configurations and speech sounds. SpeechStation analysis facilities will provide correlated displays of spectra of synthetic or natural utterances. Initial plans are to develop a complete introductory course of 16 lessons. Course material for advanced students would serve to train in research methods and teach advanced topics. A trial lesson will be demonstrated. [Work supported by NIDOCD.]
Journal of the Acoustical Society of America | 1992
Robert Berkovitz; J. M. Pickett
A course consisting of computer‐based lessons is under development to teach principles of speech sound production and acoustic patterns of phonetic features. The course employs an IBM‐compatible personal computer and a highly flexible graphic format, and makes extensive use of the speech synthesis and analysis capabilities of the SpeechStation system. The teaching approach will be demonstrated and made available for hands‐on, interactive use with lessons under development. [Work supported by the National Institute on Deafness and Other Communication Disorders.]
Journal of the Acoustical Society of America | 1988
Robert Berkovitz; Kenneth N. Stevens
An earlier publication [Stevens et al., J. Acoust. Soc. Am. 81, 470–484 (1987)] described a procedure for determining the absolute sound pressure at the medial end of the ear canal at frequencies in the range 8–20 kHz when a sound source is coupled to the ear through a lossy tube. The present paper reports comparisons of estimated and measured apex sound pressures for various ear‐canal models. Agreement was within 1–2 dB for simplified models with rigid perpendicular and oblique terminations. Changes in length over a range similar to that of human ear canals had little effect on results. Similar agreement up to about 18 kHz was obtained for nonuniform ear‐canal models with constrictions to less than one‐half the cross‐sectional area of the average ear canal, and the error was slightly larger above 18 kHz. Constrictions in the models led to nonuniformities in the frequency response at the apex. Estimates of apex sound pressures for a number of human ears exhibited features similar to those observed for the...