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Showing papers on "Microphone published in 1984"


PatentDOI
TL;DR: In this paper, a hearing aid including a microphone for generating an electrical output from sounds external to a user of the hearing aid, an electrically driven receiver for emitting sound into the ear of the user, and circuitry for driving the receiver.
Abstract: A hearing aid including a microphone for generating an electrical output from sounds external to a user of the hearing aid, an electrically driven receiver for emitting sound into the ear of the user of the hearing aid, and circuitry for driving the receiver. The circuitry drives the receiver in a self-generating mode activated by a first set of signals supplied externally of the hearing aid to cause the receiver to emit sound having at least one parameter controlled by the first set of externally supplied signals and then drives the receiver in a filtering mode, activated by a second set of signals supplied externally of the hearing aid, with the output of the external microphone filtered according to filter parameters established by the second set of the externally supplied signals. Other forms of the hearing aid, apparatus for supplying the sets of signals to the hearing aid in a total system, and methods of operation are also described.

159 citations


PatentDOI
TL;DR: An electroacoustic transducer, such as a microphone, which may be integrated into a semiconductor chip and a method of fabrication, is discussed in this paper, where the sensitivity of the device is made to be an approximately linear function of sound pressure level to be compatible with amplification.
Abstract: An electroacoustic transducer, such as a microphone, which may be integrated into a semiconductor chip and a method of fabrication. The semiconductor (10) is etched to produce a membrane (11) having a sufficiently small thickness and an area so as to vibrate at audio frequencies. Electrodes (12, 17) are provided in relation to the membrane so that an electrical output signal can be derived from the audio frequencies, or vice versa, due to variable capacitance. Preferably, the sensitivity of the device is made to be an approximately linear function of sound pressure level to be compatible with amplification.

95 citations


PatentDOI
TL;DR: In this article, a pair of sensitive microphones or transducers are mounted on a user's body, spaced apart by a distance equal to one-half a wavelength of the center frequency of a range of frequencies to be emphasized.
Abstract: A pair of sensitive microphones or transducers are mounted on a user's body, spaced apart by a distance equal to one-half a wavelength of the center frequency of a range of frequencies to be emphasized. By summing the outputs of the two microphones, sound in the broadside or look direction (i.e., the direction the listener faces, the microphones being on a line perpendicular to this direction) are emphasized; sounds in the endfire or side directions are nulled or produce a substantially null response in the region of the center frequency defined by the microphone spacing. A third microphone may be added that is not equally spaced from the microphones on either side, but is spaced to provide half wavelength distances which define maximum and null responses centered at the other points within the frequency range (1-4 KHz) desirable for highly effective hearing. The summed signal from each mircophone pair is bandpass filtered. Three bandpass filters are used. The centers of their pass bands are 1200 Hz, 2250 Hz, and 3600 Hz, respectively. Thus each microphone pair and associated bandpass filter is responsible for providing a directonal receiving capability in its assigned range of frequencies. The frequency ranges are contiguous and overlap slightly. The final output is obtained by summing and amplifying the bandpass filter outputs. A good bandpass filter design is a fourth order Butterworth filter, whose center frequency can be designed to be: ##EQU1##

94 citations


PatentDOI
TL;DR: In this paper, a system insensitive to nonspeech sounds utilizes a pair or spatially separated microphones (101, 102) to obtain the direction of origin of speech signals from a common sound source.
Abstract: A system insensitive to nonspeech sounds utilizes a pair or spatially separated microphones (101, 102) to obtain the direction of origin of speech signals from a common sound source. The speech signal from each microphone is transformed (330, 340, 350, 355, 360, 365, 370, 375) into a pulse representative signal having a rapid increase responsive to pitch peaks of energy from the sound source. The cross correlation of these pulses accurately reflects the phase relationship between the speech signals arriving at the microphones. The cross correlation is implemented (450, 460) as time interval histograms which are periodically read to identify the direction of the common sound source.

87 citations


Journal ArticleDOI
TL;DR: In this article, a near-field microphone array and a single-sensor wedge-shaped hot-film probe were used to investigate shock noise associated with unheated supersonic jets.
Abstract: Shock noise associated with unheated supersonic jets was investigated using a near-field microphone array and a single-sensor wedge-shaped hot-film probe. Both over- and underexpanded cases were investigated using Mach 1.45 and 1.99 convergent-divergent nozzles. Correlation measurements through each shock cell of a single underexpanded case with the Mach 1.45 nozzle were obtained between the hot-film probe and the microphone array. The results of the hot-film/near-field microphone correlations show general agreement with certain theoretical models as to the location for shock noise production, and provide evidence for the existence of some large-scale flow structure that collectively interacts and phases the motion of the downstream shocks. The nearfield microphone correlations demonstrate that downstream shocks dominate shock noise production and suggest the existence of a Doppler effect in the near field of the sources. In addition, broadband shock noise is found to propagate at small angles to the jet axis.

86 citations



Journal ArticleDOI
TL;DR: An acoustic reflectometer is designed, built, and tested to overcome inadequate diagnostic methods for otitis media with effusion and is reliable independent of age, crying, cerumen, and lack of cooperation from the child.

70 citations


PatentDOI
TL;DR: In this article, an active sound control system is described in which allowance is made in a relatively uncomplicated circuit for acoustic coupling between a sound generating system for generating a cancelling sound wave and a detector for sensing a sound wave to be cancelled.
Abstract: An active sound control system is described in which allowance is made in a relatively uncomplicated circuit for acoustic coupling between a sound generating system for generating a cancelling sound wave and a detector for sensing a sound wave to be cancelled Unwanted sound from a source is detected by a microphone and cancelled by sound from a speaker connected by way of an amplifier to the microphone The amplifier has a feedback processing system with a transfer function which takes account of acoustic feedback between the speaker and the microphone in deriving, with the amplifier, a signal to drive the speaker

69 citations


Patent
22 Jun 1984
TL;DR: In this paper, a read-only memory (ROM) is used to compensate for the microphone's nonlinear frequency response characteristic during their integration into a single unit, and the ROM is thus programmed to provide a frequency response complementary to that exhibited by the microphone.
Abstract: A microphone is provided with a read only memory (ROM) which is programmed to compensate for the microphone's nonlinear frequency response characteristic during their integration into a single unit. The ROM is thus programmed to provide a frequency response complementary to that exhibited by the microphone. The microphone may have a relatively poor frequency response over the audio spectrum, i.e., 20 Hz-20 kHz, but with the ROM serving to calibrate the microphone, the combination exhibits a relatively flat frequency response over this frequency range. Intended for use in a portable, battery operated, microprocessor-controlled audio spectrum analyzer, the integrated microphone and ROM combination is coupled to a CMOS random access memory (RAM) in which the calibration contents of the ROM are stored. During normal operation, the microprocessor reads the microphone calibration data from the thus programmed RAM in order to conserve battery power, but may read this data from the ROM when necessary. This permits an inexpensive microphone to be used in a high performance audio spectrum analyzer resulting in a flat frequency response.

65 citations


PatentDOI
TL;DR: In this paper, a belt-like diaphragm whose both ends are fixed to a casing with one end open was used for detecting voice vibrations by being contacted to the buccal region or the mastoid of the user.
Abstract: A vibration-detecting-type microphone for detecting voice vibrations by being contacted to the buccal region or the mastoid of the temporal region of a user. This microphone has a belt-like diaphragm whose both ends are fixed to a casing with one end open, a piezoelectric element installed on the rear central part of this diaphragm, and a vibration pickup situated on the external surface of this diaphragm and designed to be contacted with the human body. The rear surface of the vibration pickup has at least one pair of sensing elements, and these sensing elements are located in such a way that they will not oppose the piezoelectric element through the diaphragm. This arrangement makes it possible to select and set the resonance frequency of the diaphragm and microphone, as desired. In addition, since the sensing elements of the vibration pickup are supported at two points or more by the diaphragm, the concentration of stress into the diaphragm as well as the damaging of the piezoelectric element can be prevented.

65 citations


PatentDOI
TL;DR: In this paper, the similarity signal is processed with reference to a reference voltage which varies with the setting of the manual volume control of the amplifier interposed ahead of the loudspeaker.
Abstract: A microphone picks up a signal in response to both the acoustic output of the speaker and the ambient noise at the loudspeaker location, for example, in an automobile The microphone output signal and the signal which is to be provided to the loudspeaker with variable amplification are each subjected to envelope detection The two envelope curve signals are compared to produce a signal representative of their similarity, the similarity being high when ambient noise is low or absent The similarity signal is then processed with reference to a reference voltage which varies with the setting of the manual volume control of the amplifier interposed ahead of the loudspeaker That amplifier has an automatic volume control input to which a control signal generated in response to the envelope similarity signal is applied to raise the amplification and the loudspeaker output level when ambient noise, detected by the similarity of the envelope signals, is present or increases The system and method have the advantage that special calibration of the system in place of use, for proper balancing of the microphone output against the useful signal is made unnecessary The envelope signal comparison is simply performed by comparing the respective signs of the slopes of the respective envelope curves to supply a first output value when the signs are identical and a second output value when they are opposite The output values are weighted by a factor dependent upon signal dynamics (derived from average value and dynamic range) to reduce the larger output value and at the same time increase the smaller one by a dynamic factor The succession of weighted output values thus produced is integrated to produce the similarity signal

Journal ArticleDOI
TL;DR: With the materials and methods described here it is possible to achieve the same reliability for sound field testing as for audiometry under earphones.
Abstract: Comprehensive recommendations are presented for conducting sound field audiometry with frequency specific stimuli. These recommendations are primarily based on a series of investigations by the authors. The rationale for each recommendation is presented, together with a brief overview of the supporting research. The preferred stimuli are frequency modulation tones, triangularly or sinusoidally modulated at a rate of about 20 Hz, or suitably generated narrow bands of noise. The optimal bandwidths of the stimuli, expressed in percentages of the center frequency, vary with frequency. Stimuli suitable for most purposes have bandwidths ranging from about 30% at 0.25 kHz to about 10% at 4 kHz. Stimuli having narrower or broader bandwidths are desirable for some special purposes. The test room should be as nonreverberant as possible and the subject should be seated on an adjustable height chair with headrest. The control microphone method of calibration is preferred but a method is also presented for carrying out the traditional precalibration procedure. The SPL of the complex stimulus should be taken as the peak deflections on a sound level meter set to "RMS-FAST." A conversion table is presented which allows thresholds obtained in the sound field to be expressed as dB HTL. With the materials and methods described here it is possible to achieve the same reliability for sound field testing as for audiometry under earphones.

PatentDOI
TL;DR: In this paper, a self-contained electronic anti-snoring device is adapted to be worn in the outer ear or attached to the ear or earbud, and means are also provided to vary the amplitude and duration of the audio stimulus with successive snores to obviate habituation.
Abstract: A compact self-contained electronic anti-snoring device is adapted to be worn in the outer ear or attached thereto. It comprises a miniature microphone for detecting snoring sounds and means responsive to the detection of snoring sounds for generating an aversive audio signal. The aversive audio stimulus is emitted via a speaker in the user's ear. Preferably, a combined microphone/speaker is used and the snoring sounds are detected via the auditory canal. Means are also provided to vary the amplitude and duration of the audio stimulus with successive snores to obviate habituation. A counter is also provided to record the number of snores during a sleeping period and provide an indication of the effectiveness of the device.

Patent
26 Jun 1984
TL;DR: In this paper, a video camera with a stereo microphone is mounted on a main body in a reverse posture, and the top/bottom portions of picture information of a video signal supplied to the electronic view finder and also exchanges the L and R information of stereophonic signals supplied to a stereo recording apparatus.
Abstract: A video camera apparatus of the present invention has a special switch circuit which, when an electronic view finder with a stereo microphone is mounted on a video camera main body in a reverse posture, electrically reverses the top/bottom portions of picture information of a video signal supplied to the electronic view finder and also exchanges the L and R information of stereophonic signals supplied to a stereo recording apparatus. The signal selection of stereophonic signals from the stereo microphone and that of top/bottom-reversed and top/bottom-nonreversed video signals from a video camera main body are respectively changed in accordance with the mounting posture of the electronic view finder. Therefore, a proper stereophonic recording can be effected both in the normal and reverse posture photographing.

Journal ArticleDOI
TL;DR: The present design was developed as an alternative to the fragile aluminum-ribbon microphones used previously and has the advantages of low cost, small size, robustness and a broad frequency response.
Abstract: Conventional particle velocity microphones (see Olson, 1957) are not readily available, are expensive, fragile and large, so they are not commonly used in biological research, though their use in one situation where the source is highly reactive, the recording of Drosophila song, is now standard practice. Here, the particle velocity produced by the fly is far louder than the sound pressure, so high quality recordings can be made with minimal sound insulation (see Bennet-Clark, 1971; Bennet-Clark, Leroy & Tsacas, 1980). The microphones can also be used to localize sources of echoes and to measure other reactive conditions. The present design was developed as an alternative to the fragile aluminum-ribbon microphones used previously (Bennet-Clark, 1973) and has the advantages of low cost, small size, robustness and a broad frequency response. The transducer is an electret membrane open on both sides to the sound wave. The resonant frequency of the electret unit is at the top of the practical frequency range of the microphone (16—17 kHz). Below this, the output is proportional to the driving force, the pressure gradient, so rises 6dB per octave and leads by 90 in phase compared with the particle velocity (see Michelsen & Nocke, 1974). The microphone has three major components, a pressure gradient transducer, an FET impedance converter in the microphone head and an integrating amplifier to correct the frequency and phase response of the transducer. The electret unit is Radio Shack type 270-090 (Fig. 1) (Tandy Corporation; products are available world-wide). Retain the metal-coated electret membrane on its supporting ring; the 40/im thick plastic spacer washer, the perforated fixed electrode and the FET (see Fig. 1). The membrane is reassembled so as to be open to the sound wave on both sides. Dimensions of a brass housing and PTFE or Delrin (Acetal) insulator are shown in Fig. 2A. Assembly is shown in Fig. 2B. The electret and fixed electrode are separated by the 40 /xm spacer washer and the fixed electrode is separated from the housing by the insulator. The edges of the brass housing can then be crimped over to secure the components in position. If the electret is heated, it may discharge. The FET amplifier should be built close to the transducer. A suitable layout, which incorporates the 1 • 1 k£2 FET load resistor, is shown in Fig. 2C and the circuit diagram is shown in Fig. 3. In this layout, response varies as the cosine of the angle from the

Proceedings ArticleDOI
19 Mar 1984
TL;DR: An electronic controller based on a digital implementation of transversal filters using a modification of the Widrow-Hoff Least-Mean-Squared Algorithm for active noise reduction is described.
Abstract: The active reduction of noise is an application of the principle of superposition in which an unwanted noise signal is detected by a microphone and processed by an electronic controller to produce an equal amplitude, 180° out-of-phase cancellation signal. The signal is then appropriately amplified and injected back into the space by a loudspeaker. This paper describes an electronic controller based on a digital implementation of transversal filters using a modification of the Widrow-Hoff Least-Mean-Squared Algorithm. In the case of active noise reduction there is no signal to enhance and all of the detected input must be cancelled. Further, the time required for the physical acoustic wave must be taken into consideration for the system to work.

Journal ArticleDOI
TL;DR: The acceleration response of microphones applied to the chest wall is determined, which is both of interest for standardised and quantitative phonocardiography.
Abstract: The determination of the effect of loading of the chest wall exerted by a contact phonocardiographic microphone, has been attempted in earlier investigations by the measurement of the mechanical properties of the microphone and the chest wall in combination with mechanical models describing the coupling phenomenon (method 1). For computing the acceleration response of the microphone applied to the chest wall a number of unverified assumptions were made. In the paper a method is described for experimental determination of the coupling characteristic. A specially designed ultralight pick-up (1 g) was applied for recording the quasi-unloaded vibrations and the average amplitude spectrum was calculated. The procedure was repeated at the same site, the chest wall successively being loaded with different masses and two commercial microphones (method 2). Logarithmic transfer functions of the loaded to unloaded situations have been computed. The experiments were performed on six healthy persons. The results of method 1 appeared to be very similar to those obtained with method 2 when a Verburg model was assumed. In this manner we determined the acceleration response of microphones applied to the chest wall, which is both of interest for standardised and quantitative phonocardiography.

Journal ArticleDOI
TL;DR: Using an electronic stethoscope, the authors have attempted noninvasive detection of intracranial aneurysms, arteriovenous malformations (AVM's), and carotid cavernous fistulas in 45 patients, finding neither spikes nor bruits were demonstrable in three patients with brain tumors or in seven patients without intrac cranial vascular lesions.
Abstract: ✓ Using an electronic stethoscope, the authors have attempted noninvasive detection of intracranial aneurysms, arteriovenous malformations (AVM's), and carotid cavernous fistulas in 45 patients. A microphone of older design and a newly designed horn-coupled probe microphone were used to record the sound signals emanating from the cranium. A trigger pulse recorded by another microphone placed over the carotid area or the precordium was used to time the intracranial signals. The sound signals were converted to electrical signals, amplified, filtered, and analyzed using fast Fourier transformation to give plots of amplitude versus frequency of the signals. A spike at a certain frequency or a bruit over a band of frequencies was considered a positive finding. The records of 18 of the patients were not satisfactory for analysis, mainly due to external noise interference. Eight of 11 aneurysm patients with satisfactory recordings emitted resonant spikes, turbulent bruits, or combinations of the two. The other t...

PatentDOI
TL;DR: In this article, an omnidirectional microphone system with a range finder for determining the distance between the microphone and a speaker is provided. And in further embodiments, there are also provided a multiplicity of spaced range finders for determining exact position of a speaker or speakers relative to a microphone or several microphones.
Abstract: A microphone ranging system for providing improved reproduced speech by compensating for the position of a speaker relative to the microphone system. An omnidirectional microphone system with a range finder for determining the distance between the omnidirectional microphone and a speaker is provided. In further embodiments, there are also provided a multiplicity of spaced range finders for determining the exact position of a speaker or speakers relative to a microphone or several microphones. Highly directional microphones are used which are trained on the speaker or speakers and follow them as they move about a room. A tracking system is controlled through signals which originate in a control circuit which is controlled by the range finders.

PatentDOI
TL;DR: In this paper, a hearing protection device consisting of a headset having opposite earmuff and/or earplug assemblies resiliently connected by a headband assembly including a normally open power switch closed when the headset assembly is put on the user's head.
Abstract: A hearing protective device consisting of a headset having opposite earmuff and/or earplug assemblies resiliently connected by a headband assembly including a normally open power switch closed when the headband assembly is put on the user's head. The switch controls the energizing of the electrical circuitry of the device. Each earmuff and/or earplug assembly has an outwardly-facing microphone and an inwardly-facing sound reproducer, connected to circuitry defining respective stereo channels, each channel including preamplifier circuitry and a respective power amplifier drivingly connected to one of the sound reproducers. The circuitry also includes a hyper AGC circuit which receives and sums signals from the amplifier chain and derives attenuation signals therefrom which are fed back to the amplifier circuitry and which reduces the gain of this circuitry when the input sound exceeds a certain level. This causes the inputs to the power amplifiers to fall when the input to the preamplifiers increases beyond a certain level. A balanced attenuation circuit arrangement is employed which reduces the size of too-large sound waves without altering their shape, whereby the volumes of the two stereo channels are maintained in the proper relation to each other to preserve binaural hearing.

PatentDOI
TL;DR: In this paper, a multi-band amplification circuit for hearing aids performs independent amplification of different frequency bands of the signal generated by a microphone of the hearing aid, to match the loss of hearing of a user of hearing aid.
Abstract: An amplification circuit for hearing aids performs independent amplification of different frequency bands of the signal generated by a microphone of the hearing aid, to match the loss of hearing of a user of the hearing aid. Prior to this multi-band amplification, however, the microphone output signal is passed through an optional filter having a characteristic which is the inverse of the spectrum of long term speech, and through a signal compressor. The signals from the multi-band amplification are combined using an adder and the combined signal is modified by a control signal derived from the envelope of the signal generated by the microphone. The result of this processing is to cause the range of spectral shapes present in received signals to be enlarged or exaggerated in the output signal, but to cause the range of overall levels present in the received signal to be decreased in the output signal. The processing thus increases both signal comfort and signal discriminability for the hearing impaired user of the hearing aid.

Patent
22 Mar 1984
TL;DR: In this paper, an abnormal noise detector for detecting abnormal noises produced by a gear unit with an eccentrically mounted gear is presented, which includes a microphone, a digital frequency analyzer and a central processing unit.
Abstract: An abnormal noise detector for detecting abnormal noises produced by a gear unit with an eccentrically mounted gear is disclosed. A first embodiment comprises a microphone placed in the vicinity of a gear unit to be tested, a bandpass filter connected to the microphone which passes only a band of frequencies centered around an integral multiple of the normal fundamental frequency of the noise produced by the meshing of the gears of the gear unit, an amplitude detector which detects the amplitude of the signal from the bandpass filter, a frequency deviation detector which detects the frequency deviation of the signal from the bandpass filter, and recorders for recording the values of the signals from the amplitude detector and the frequency deviation detector as a function of time. A second embodiment comprises a microphone, a digital frequency analyzer and a central processing unit for determining whether abnormalities exist in the data produced by the digital frequency analyzer.

PatentDOI
TL;DR: In this article, a pair of parallel-connected directional microphone elements are mounted in opposite directions and mounted in a parabolic reflector, and the microphone element output signals are separated into infrasonic, sonic, and ultrasonic channels by filters.
Abstract: Apparatus for enhancing human hearing includes a pair of parallel-connected directional microphone elements responsive to frequencies ranging from below the audible frequency range to above the audible frequency range, the microphone elements facing in opposite directions and mounted in a parabolic reflector. The microphone element output signals are separated into infrasonic, sonic, and ultrasonic channels by filters. The infrasonic and ultrasonic channel signals are processed by a frequency heterodyning process to signals having frequencies within the range of human hearing. The channel signals are then selectively combined, amplified, and supplied to an earphone to render audible sounds outside the range of human hearing and aid in the detection, location, and classification of events of interest.

PatentDOI
TL;DR: In this paper, an earphone characteristic measuring device consisting of an acoustic coupler having an acoustic tube simulated to an external auditory canal, a microphone mounted at the end of the first acoustic tube for picking up sound pressure information, and a characteristic calculation circuit was used to transform the earphone characteristics of the coupler to those of a real ear.
Abstract: An earphone characteristic measuring device comprises an acoustic coupler having an acoustic tube simulated to an external auditory canal in which an earphone under measurement is to be inserted and an acoustic tube of a smaller diameter having an acoustic impedance of approximately 320 ohms connected to an end of the first acoustic tube, a sound source for emitting an impulse sound to the acoustic coupler, a microphone mounted at the end of the first acoustic tube for picking up sound pressure information and a characteristic calculation circuit for transforming an earphone characteristic of the acoustic coupler to an earphone characteristic of a real ear based on an input impedance of the acoustic coupler viewed from an end of the earphone inserted in the acoustic coupler and an input impedance of the real ear represented by a sum of an eardrum impedance of the real ear and an external auditory canal volume of the real ear, stored in a memory in response to the sound pressure information from the microphone. The use of the acoustic coupler of a simple structure facilitates the measurement of a vent characteristic of the earphone and an insertion gain and improves reliability of the measurement.

Patent
22 Jun 1984
TL;DR: In this paper, the detection of low-flying aircraft is accomplished using both a microphone and geophone as sensors and a signal processing system for measuring the correlation between the seismic and acoustic signals.
Abstract: The detection of low-flying aircraft is accomplished using both a microphone and geophone as sensors and a signal processing system for measuring the correlation between the seismic and acoustic signals. The signal processing system contains two amplification and band-limiting circuits, a delay circuit, an adaptive noise cancelling circuit, two signal smoothing circuits and a comparison circuit. The two amplification and bandlimiting circuits enhance a selected portion of the seismic and acoustic signals received by the sensors. The optional delay circuit is applied to the input acoustic signal, allowing the user the option of sensing or rejecting the detection of jet aircraft. The adaptive noise cancelling circuit generates an error signal by subtracting the signal components which are correlated between the seismic and acoustic signals from the input seismic signal. The first signal smoothing circuit improves the quality of the input seismic signal for comparison. The second signal smoothing circuit improves the quality of the error signal generated by the adaptive noise cancelling circuit. Finally, the comparison signal compares the smoothed seismic signal with the error signal. Low-flying aircraft have seismic and acoustic signatures which are highly correlated while other sources (wind, seismic events and motor vehicles) have little correlation. Therefore, if the error signal resembles the seismic signal there occured little correlation and the causing the event was not an aircraft. Dissimilar error signals and seismic signals occur when aircraft are detected, an event which is indicated by the output signal from the comparison circuit.

Patent
23 Mar 1984
TL;DR: In this article, a microphone designed for use on location provided with amplification to produce a signal output level suitable for using on conventional studio cables and utilizing the Phantom power available on such studio cables, the microphone having a light emitting diode responsive to elevation of the potential of Phantom power on the cable, but not the normal potential thereof, to indicate the presence of a live microphone, also being provided with a backup battery for use in the event Phantom power is not available.
Abstract: A microphone designed for use on location provided with amplification to produce a signal output level suitable for use on conventional studio cables and utilizing the Phantom power available on such studio cables, the microphone having a light emitting diode responsive to elevation of the potential of the Phantom power on the cable, but not the normal potential thereof, to indicate the presence of a live microphone, the microphone also being provided with a backup battery for use in the event Phantom power is not available, and the microphone being provided with a free running multivibrator coupled to the light emitting diode and having a repetition rate proportional to the potential of the backup battery to give an indication of battery voltage.

Patent
Kinichi Uno1, Tohru Nakahara1
16 Apr 1984
TL;DR: In this paper, the authors propose a circuit for preventing howling responsive to an acoustic feedback between a microphone and a loudspeaker in a hands free telephone or the like, where a variable loss circuit and a level detector circuit are used to compare the level of a signal picked-up by a microphone with a level received from another party.
Abstract: The invention provides a circuit for preventing howling responsive to an acoustic feedback between a microphone and a loudspeaker in a hands free telephone or the like. Both a transmit path and a receive path include a variable loss circuit and a level detector circuit which compares the level of a signal picked-up by a microphone with a level of a signal received from another party. Responsive to the comparison, a normal gain is set in one of the communication paths at a level which is higher than the signal level in the path. A loss is inserted in the other communication path. A control circuit compares outputs of the level detector circuits to control the losses of the variable loss circuits. To measure a loss due to the acoustic coupling and to measure a reverberation time, a manual switch is operated to drive a microprocessor type control circuit responsive to an in-band noise which produces a voice-band noise signal from the loudspeaker. The acoustic coupling loss and reverberation time are found by computation and then stored. Signals transmitted and received during conversation are processed to control the variable loss circuits of the communication paths, on a basis of stored signal levels.

Journal ArticleDOI
TL;DR: Improve in SNR as a result of the AMNOR system by more than 15 dB in the frequency range of 300–3300 Hz and Superiority over the conventional LMS criterion for noise reduction of speech signals was confirmed by subjective preference tests.
Abstract: A new adaptive microphone‐array system for noise reduction (AMNOR system) is introduced. The AMNOR system detects the arrival directions of noise and forms a directional pattern possessing low sensitivities to those directions using a digital filtering technique. With pre‐informed data on the direction of desired signal source, the AMNOR system learns the noise arrival directions previously. Optimum filter coefficients are determined based on a new criterion, whose concept is to minimize output noise power while maintaining the degradation in the frequency response to the desired signal below the predetermined value. An algorithm for calculating the optimum filter is proposed. To confirm the effectiveness of the AMNOR system, experiments on the noise reduction processing were carried out in a room with a reverberation time of 0.4 s. A small circular microphone array (radius of 8.5 cm) with four microphone elements and four FIR filters with 16 taps were used. The convergence time of the algorithm at this experimental condition was 0.3 s. Improvement in SNR as a result of the AMNOR system by more than 15 dB in the frequency range of 300–3300 Hz. Superiority over the conventional LMS criterion for noise reduction of speech signals was also confirmed by subjective preference tests.

Journal Article
TL;DR: Les melangeurs automatiques sont souvent employes dans les systemes sonores a microphone multiple pour mettre en marche les microphones immediatement en service as mentioned in this paper.
Abstract: Les melangeurs automatiques sont souvent employes dans les systemes sonores a microphone multiple pour mettre en marche les microphones immediatement en service

PatentDOI
Friedrich Harless1
TL;DR: In this article, the authors describe a hearing aid that has a housing containing components such as a microphone, an amplifier and an earphone, to which a terminal having contacts for connection to a signal line is applied.
Abstract: A hearing aid apparatus has a housing containing components such as a microphone, an amplifier and an earphone,and to which a terminal having contacts for connection to a signal line is applied. The terminal also has contacts for derivation of signals from the microphone so that a conventionally operating hearing aid is achieved which also contains an audio output in addition to an audio input. The hearing aid apparatus is thus particularly suitable for use as a talk/listen set in aural training systems.