Showing papers on "Microphone published in 1987"

Ultrasound position input device

Abstract: This invention is an input device, or stylus, for entering hand drawn forms into a computer comprising a writing instrument, a pressure switch for determining whether the instrument is in contact with the writing surface, an acoustic transmitter for triangulating the position of the stylus on the surface, and a wireless transmitter for transmitting data and timing information to the computer. In operation, the stylus transmits an infra red signal which the system receives immediately, and an ultra sound pulse which two microphones receive after a delay which is a function of the speed of sound and the distance of the stylus from each microphone. From this information the system can calculate the position of the stylus. Switches for indicating functions are mounted on the stylus. Multiple stylusses can be used, each transmitting a distinctive identification code so that the system can determine which stylus is the signal source.

Card with built-in record/playback capability

Abstract: Card or postal media which comprises a storage member to retain audio information picked up by a microphone, a sound generating member including a speaker, a mode selection signal producing member to allow either record or playback to be selected, a controller performing record or playback according to the mode selection signal from the mode selection signal producing member; converting the audio signal from the microphone from analog to digital and storing it in the storage member when the record mode is selected; retrieving stored information from the storage member, converting it from digital to analog, and outputting the analog signal to the sound generating member when the playback mode is selected, and a card board on which the microphone, speaker, and the other members are mounted.

IC-Processed piezoelectric microphone

Abstract: A miniature diaphragm pressure transducer having sensitivity to acoustic signals at the level of conversational speech has been fabricated by combining micromachining procedures (to produce a thin silicon-nitride diaphragm) with ZnO thin-film processing. The sensor consists of a patterned ZnO layer (which acts as a piezoelectric transducer) deposited on a thin square micromachined diaphragm made of LPCVD silicon nitride. The diaphragm, 2 µm in thickness, is the thinnest yet reported for a piezoelectric readout structure of relatively large area (3 × 3 mm2). The transducer shows an unamplified response of roughly 50 µV/µbar when excited by sound waves at 1 kHz with the variation of the sensitivity from 20 Hz to 4 kHz being approximately 9 dB. These results are obtained using a 0.1-mm-wide annular pattern that measures 3.6 mm in circumference.

Noise reduction processing arrangement for microphone arrays

Abstract: A sound processing arrangement couples sound from a prescribed environment through a fixed microphone array to a signal processing arrangement having a specifiable preferred sound source location. The microphone pickup signals are combined with a set of weighting signals to adjust the directional response pattern in successive analysis time intervals. The weighting signals are modified in each analysis time interval so that the total acoustic signal power of the signal processing arrangement output signal is decreased toward a minimum while substantially unity power transfer of sound signals from said preferred location is maintained at all frequencies over a prescribed frequency range. In this way, the preferred source location is in the main beam while unwanted sound source locations are at the null points of the adjusted directional response pattern.

Multi-channel cochlear implant system

Abstract: A method and system is described for electrically stimulating the auditory nerve with multiple channels of audio information. Audio information is transmitted from an external microphone to a speech processor through a single transcutaneous path to an implanted receiving device. Power is transmitted through a second transcutaneous path to power the implanted device.

Emergency signal warning system

Abstract: A siren detection system is particularly suited for installation in an automobile for providing a warning to the driver when an emergency vehicle is in the vicinity of the automobile. The system includes a microphone mounted on the automobile. The output of the microphone is supplied to a bank of band-pass filters, each tuned to a different adjacent portion of the frequency range through which the frequency of a siren varies. The output of the filters are sampled by an averaging and selection circuit which produces a single output corresponding to the central one of any of the filters producing an output during a preestablished time interval. This output then is compared with the previous output (which indicates the previously selected band-pass frequency) to operate an up/down counter. Whenever successive counts of a preestablished number in the same direction occur, the output of the counter enables an alarm to indicate the presence of a siren operated by an emergency vehicle in the vicinity of the automobile. The system also includes circuits for enhancing noise rejection in the same frequency range.

Method and apparatus for attentuating external origin noise reaching the eardrum, and for improving intelligibility of electro-acoustic communications

Abstract: The invention relates to a method and to apparatus for attenuating externally originating noise reaching the eardrum while still enabling communication via an electro-acoustic path. Apparatus according to the invention comprises passive attenuation means disposed about each ear and delimiting a cavity (10). In addition it includes active attenuation means comprising a loudspeaker (6) placed inside the cavity (10) and a microphone (8) placed in the external ear duct or at the inlet thereto, said loudspeaker and microphone being interconnected by a constant gain amplifier (11) and an active analog filter (12) of the polynomial type, with the passive components thereof being designed to provide a given transfer function. One application lies in the construction of protective headsets fitted with incorporated loudspeakers enabling electro-acoustic communication.

Cleanable in-the-ear electroacoustic transducer

Abstract: A hearing aid including a microphone, an adjustable amplifier, and a transducer comprising an acoustic driver driving a diaphragm disposed within an acoustic chamber, all mounted in an in-the-ear housing having a sound outlet passage leading from the acoustic chamber into the user's ear canal, is provided with a cleaning passage that is accessible from the outside of the housing and that connects to the inner end of the sound outlet passage, through a portion of the acoustic chamber. Throughout internal cleaning is effected by pumping a solvent through the continuous conduit formed by the cleaning passage, the acoustic chamber, and the sound outlet passage, without disassembly of the hearing aid.

Motorcycle stereo audio system with VOX intercom

Abstract: An improvement to motorcycle intercom communications between the motorcycle driver and passenger providing for stereo listening in each of both right and left helmet earphones wherein intercom communications may be had through microphones always activated located proximate the mouth of both driver and passenger where when initiating speaking over the microphone, the stereo output of one side of the earphones is reduced by approximately one/half while the stereo output of the other side earphones is terminated completely and the intercom conversation is heard in full over that single side of earphones. Upon the termination of the intercom conversation, the invention circuitry returns both sides of earphones to the pre-existing level of stereo output. The above is accomplished by dividing all sounds which come in over the microphone into a low and high frequency range separated by an audio speaking range wherein the outputs of the low and high frequency range filters are constantly sampled and compared with the energy output of the audio speaking range of frequencies. When the energy in the speaking range of frequencies exceeds the energy in the low and high frequency range, such is indicative that a party is speaking into the microphone and the circuit automatically terminates the output of one earphone, reduces the other by half, and injects the intercom on the earphone whose stereo had been terminated.

Multimicrophone adaptive beamforming for interference reduction in hearing aids.

Abstract: To reduce interference in monaural hearing aids from sound sources that are spatially separated from a target source, we are investigating methods for combining information from multiple microphones. In this paper, we describe an adaptive beamforming method that functions to preserve target signals arriving from straight-ahead of a microphone array while minimizing output power from off-axis interference sources. In a preliminary evaluation of a two-microphone system, sentence intelligibility tests were administered to normal-hearing subjects using processed and unprocessed materials from simulated environments in which the target was on-axis, the interference (speech babble) was 45 degrees off-axis, and the reverberation mimicked that of a living room, a conference room, and anechoic space. Compared to listening through a single microphone, the two-microphone beamformer reduced the target-to-interference ratio required to achieve 50 percent keyword intelligibility by 30, 14, and 0 dB in the anechoic, living-room, and conference-room conditions, respectively. The corresponding improvements over binaural listening (one microphone to each ear) were 24, 9, and 0 dB. Further tests in the living-room environment using the same beam-forming system but with filter impulse responses shortened by a factor of four (which would decrease the adaptation time by a factor of four) decreased the improvement by 5 dB. These results are sufficiently encouraging to warrant further tests involving more realistic reverberant conditions, multiple sources of interference, and time-varying acoustic environments.

Speech recognition system

Abstract: A plurality of microphones are disposed on a body to detect the speech of a speaker. First, second and third microphones may respectively detect the sounds emanating from the speaker's mouth, nose and throat and produce signals representing such sounds. A fourth microphone may detect the fricative and plosive sounds emanating from the speaker's mouth and produce signals representing such sounds. The signals from the different microphones are compared to allow the discrimination of certain speech sounds. For example, a high amplitude of the signal from the nose microphone relative to that from the mouth microphone indicates that a nasal sound such as m, n, or ng was spoken. Identifying signals are provided to the speech recognition system to aid in identifying the speech sounds at each instance. The identifying signals can also select a microphone whose signal can be passed on to the recognition system in its entirety. Signals may also be provided to identify that spoken words such as "paragraph" or "comma" are actually directions controlling the form, rather than the content, of the speech by the speaker. The selected signals, the identifying or classifying signals and the signals representing directions may be recovered by the system of this invention. The selected and identifying signals may be processed to detect syllables of speech and the syllables may be classified into phrases or sentences. The result may then be converted to a printed form representing the speech or utilized in the operation of another device.

Some analysis of microphone arrays for speech data acquisition

Abstract: Speech data for recognition, talker verification, or recording are typically acquired using a head-mounted or hand-held microphone. These devices can be very inconvenient or provide a poor signal-to-noise ratio. A microphone array has the potential for surmounting both of these problems. Here, results on optimal spacing and gain for practical linear arrays are derived under the assumption that the desired signal and intrusive speech may be accurately modeled by plane waves. Several sets of curves are presented from the solution of constrained and unconstrained multidimensional nonlinear equations derived from the theory. The curves suggest optimal spacing and gains for linear microphone arrays for speech acquisition.

Aircraft cabin noise control apparatus

Abstract: In propeller of fan driven aircraft, cabin noise levels may be reduced by adjustment of the phase relationship between a reference propeller or fan and some or all of the other propellers or fans. An aircraft cabin (1) contains four microphones and two loudspeakers which form the active elements of a noise control system. The microphone outputs are fed via amplifiers to a digital signal processor (11) having an adaptation algorithm in a memory store. The processor generates an error signal which is used to adjust the synchrophase angle between the reference propeller and a synchrophased propeller, controlled by a synchrophaser. Thus the synchrophase angle is varied dynamically during flight to minimize propeller noise in the cabin over a range of flying conditions.

Noise compensation in speech recognition

Abstract: In speech recognition it is advantageous to take account of noise levels both in recognition and training. In both processes signals reaching a microphone l0 are digitised and passed through a filter bank to be separated into frequency channels. In training, a noise estimator 20 and a masker l5 are used with a recognizer l8 to prepare and store probability density functions (p.d.f.s) for each channel partially defining Markov models of words to be recognized. The p.d.f.s are derived only from input signals above noise levels but derivation is such that the whole of each p.d.f. is represented. In recognition, "distance" measurements on which recognition is based are derived for each channel. If the signal in a channel is above noise then the distance is determined. by the recognizer, from the negative logarithm of the p.d.f. but if a channel signal is below noise then the distance is determined from the negative logarithm of the cumulative distance of the p.d.f. to the noise level.

Wireless hand held microphone

Abstract: A microphone is provided in which two circuit boards are held in spaced parallel position by each being connected at one end to a supporting member and by each resting at the other end in a pair of stabilizing grooves located within the microphone head. A battery holding assembly has attached at one end the supporting member, and at the other end an antenna mounting surface onto which a coiled antenna is placed. A battery holding compartment is located between the supporting member and the antenna so that a battery held in the compartment acts as an addition to the antenna itself. A power and battery level indicator flashes on and off to warn that the charge of the battery is reaching a predetermined low point. The electronic circuitry for the indicator is also provided.

Calibration of ear canals for audiometry at high frequencies.

Abstract: 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 reponse 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.

The acoustic properties of the infant ear. A preliminary report.

Abstract: This is a preliminary report about the acoustic characteristics of the external ears of infants. A technique was developed to insert a probe tube that is attached to a miniature microphone into the external auditory canals of sleeping infants. The inlet to the microphone was positioned in the lateral half of the external auditory canal. A diffuse sound field (spectral density of approximately 45 dB SPL) was introduced. The microphone output was recorded, and its Fourier Transform was computed. Diffuse-field-to-ear canal sound pressure level transformations were determined for infants ranging in age from newborn to 37 months. Representative sound pressure level transformations are presented. These are shown to vary systematically with the age of the child. The resonance frequency of the external ear is high in the newborns and declines with age. The asymptotic value (approximately 2,700 Hz) is reached during the second year of life. These findings have potential implications for fitting hearing aids on infants and children.

Audio communications module for an office chair

Abstract: An audio communications system for an office chair provides a user with private listening of RF transmitted audio messages in an exposed environment such as an office area, without the use of headphones. An audio module, which mounts to the backrest of an office chair, includes an RF receiver and a shaped, acoustic horn loudspeaker which directs sound upwardly, behind the user's head, while minimizing lateral dispersion of the sound. In another aspect of the invention, the audio communications system includes a transceiver for two-way communication, such as telephone, and an armrest mounted microphone and keypad.

The measurement of periodic acoustic fields using photon correlation spectroscopy

Abstract: A method of measuring the velocity amplitudes of acoustic vibrations using the laser Doppler photon counting technique (photon correlation spectroscopy) is described. The results are compared with measurements made using a pressure microphone and close agreement is found. A gating technique is also described which allows time histories of the fluctuation to be obtained if periodicity is assumed.

A single-microphone-based self-adaptive filter of noise from speech and its performance evaluation.

Abstract: This paper discusses a single-microphone-based self-adaptive filter of environmental noise from speech. This filter, based on the work of Graupe (3) and of Graupe and Causey (4), has been incorporated in standard in-the-ear (ITE) and in behind-the-ear (BTE) hearing aids by several hearing aid manufacturers. Intelligibility tests by the authors and by independent researchers are presented in this paper to illustrate the filter's performance. Significant monosyllabic-word-list intelligibility improvements are shown in hearing-impaired and in normal-hearing subjects for virtually any environmental noise, including white noise, babble (interfering background conversations), cafeteria noise, high-frequency noise, and low-frequency noise at signal-to-noise ratios to below -20 dB.

Paradac wind measurement system

Abstract: This invention relates to a parametric source Doppler acoustic wind profiler for measuring atmospheric wind conditions. It includes a parametric acoustic source excited by a transmitter generating a pair of primary frequencies and radiates an intermodulation difference frequency. Echoes of the difference frequency are produced by wind induced scatter at the difference frequency which is significantly lower than the transmitted primary frequencies and detected by a single microphone or an array of microphones.

The lower limit of detectable sound pressures

Abstract: The electronic thermal noise in a classical condenser microphone and the associated preamplifier has been studied experimentally and theoretically. A new method was used to compute the electronic noise in the microphone. It is shown that the electronic noise in some microphones dominates the noise in the preamplifier. Therefore, the lowest sound pressure that may be detected by those microphones is determined by the thermal noise in the condenser microphones alone.

Adaptive expander for telephones

Abstract: An adaptive expander for telephones reduces the gain of a transmitting amplifier in proportion to the intensity of the background noise. Gain reduction occurs when the user is not speaking into the handset microphone. When the user is speaking, however, the gain of the transmitting amplifier is restored to its normal level. Noise is distinguished from speech via long-term averaging of the microphone output signal, and a circuit that precludes the magnitude of the long-term average voltage from ever exceeding the short-term average voltage. The present invention is used in full duplex arrangements where simultaneous transmission in both directions is possible rather than in half duplex arrangements, such as a speakerphone, where transmission occurs only in one direction at a time.

Use of an adaptive noise canceler as an input preprocessor for a hearing aid.

Abstract: The use of a two-channel adaptive noise canceler as a preprocessor for a hearing aid was evaluated. An omnidirectional microphone and a directional microphone were used as the inputs to the primary and reference channels, respectively, of the adaptive noise canceler. The microphones were mounted just above one ear on the head of a KEMAR mannikin. The system was found to work well under anechoic conditions, but showed only modest improvements in a moderately reverberant room (reverberation time 380 ms).

Effect of two-microphone noise reduction on speech recognition by normal-hearing listeners.

Abstract: An idealized 2-channel noise reducing adaptive filter of the type developed by Widrow requires that one channel contain noise only and that the microphones be fixed in position relative to the signal and noise sources. These conditions are unlikely to be met in a wearable hearing aid. In a typical situation, the microphones will be mounted in close proximity on a moving head in a room that is moderately reverberant. Experimental data have been obtained showing that, despite these deviations from the ideal conditions, significant improvements in speech intelligibility can be obtained using 2-channel adaptive filtering.

Tuning indicator for musical instruments

Abstract: A tuning indicator is disclosed in which the octave, note within the octave, and a tuning error is displayed for a musical tone played into a microphone. A bank of digital octave filters, a bank of digital note filters, a bank of digital cent filters operate simultaneously and in parallel to analyze the fundamental frequency of the musical tone. The filters operate by computing the autocorrelation function of the input signal and then performing a Fourier transform to obtain the frequency analysis data. An efficient and simple implementation is disclosed for the computations including the analog-to-digital signal conversion, the computation of the autocorrelation function, and the Fourier transform.

Portable recording system for telephone conversations

Abstract: A portable system for recording telephone conversations from a telephone and also for recording face-to-face conversations includes a portable, battery-powered audible sound recorder carried by the user, a microphone connected to the recorder, and a microphone support which is supported by the ear of the user for retaining the microphone between the telephone earpiece and the inner portion of the auditory meatus of the user. By locating the microphone near the auditory meatus or in the outer portion of the auditory meatus of the user, sounds of the face-to-face conversation and sounds of the telephone conversation from the earpiece are recorded on the recorder. With the recording system of the invention, there is no physical contact between the microphone and the telephone earpiece. A microphone holder is also disclosed for retaining a small microphone near or in the auditory meatus of the user. By using the microphone holder, any conventional portable audio recorder in conjunction the small microphone can be converted into a portable telephone conversation recording system or portable face-to-face conversation recording system. A combined microphone/speaker can be used for both recording sounds and for playing back the recorded sounds.

Adaptive noise reduction in aircraft communication systems

Abstract: In many military environments, such as fighter jet cockpits, the increasing use of digital communication systems has created a need for robust vocoders and speech recognition systems. However, the high level of ambient noise in such environments makes vocoders less intelligible and makes reliable speech recognition more difficult. One method of enhancing the noise-corrupted speech is adaptive noise cancellation. In previous research, this method was tested in a simulated cockpit environment, yielding impressive results. However, in new simulations, reflecting more realistic conditions, adaptive noise cancellation has been less successful. Spectral analysis of the data showed that the diffuseness of the ambient noise fieid, along With the microphone characteristics, has a significant effect on the performance of adaptive noise cancellation.