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Showing papers on "Microphone published in 1981"


Journal ArticleDOI
TL;DR: In this paper, a digital computer simulation of adaptive closed-loop control for a specific application, sound cancellation in a duct, is presented, which is an extension of Sondhi's adaptive echo canceler and Widrow's adaptive noise canceler from signal processing to control.
Abstract: Most active sound cancellation systems reported in the literature use open‐loop control, depend on near‐zero phase delay in control system elements, and require constant acoustic signal transit time from a signal pickup (microphone) to a control sound source (loudspeaker). The applicability of such systems can be significantly enhanced by using closed‐loop control. This study concerns a digital computer simulation of adaptive closed‐loop control for a specific application, sound cancellation in a duct. The key element is an extension of Sondhi’s adaptive echo canceler and Widrow’s adaptive noise canceler from signal processing to control. The adaptive algorithm is thus based on the LMS gradient search method. The simulation shows that one or more pure tones can be canceled down to the computer bit noise level (−120 dB). In the presence of additive white noise, pure tones can be canceled to at least 10 dB below the noise spectrum level for SNR’s down to at least 0 dB. The underlying theory implies that the algorithm allows tracking tones with amplitudes and frequencies that change more slowly with time than the adaptive filter adaptation rate. The theory implies also that the method can cancel narrow‐band sound in the presence of spectrally overlapping broadband sound. The method can be applied more widely, particularly to control systems that involve transport delay.

382 citations


Journal ArticleDOI
TL;DR: In this article, a phase locking mechanism was proposed to explain the difference between real-time and time-averaged emissions in the human cochlea, which can be used to measure tuning curves or to study the distortion product 2f1-f2.
Abstract: Cochlear acoustic emissions were recorded with a sensitive microphone from within the ear canal of normal hearing subjects. Frequency spectra of these acoustic emissions were obtained with two different procedures: real‐time recording or calculation of the spectrum of time‐averaged emissions. The two procedures give different input–output curves for click‐evoked acoustic emissions. A phase‐locking mechanism is proposed to explain this difference. It is shown that the real‐time spectrum recording procedure can be used to measure tuning curves or to study the distortion product 2f1–f2. Experimental results indicate that sharply tuned emission generators are present in the human cochlea.

75 citations


Journal ArticleDOI
TL;DR: In this article, the authors presented an error analysis of the spectral estimates used in these techniques and showed that minimum bias error can be achieved by using a small bandwidth in estimating the spectra and by locating the microphones close to the sample.
Abstract: Several methods have been proposed using multiple‐point pressure measurement of random sound fields in ducts to determine acoustic properties of materials and systems. This paper presents an error analysis of the spectral estimates used in these techniques. Expressions for the normal acoustic absorption coefficient and impedance are derived for a random sound field in a duct. Theory is developed to determine the bias and random errors in estimating the spectral density function for plane‐wave propagation in the duct. A bivariate stochastic process has been employed to model the acoustic system. Experimental and theoretical calculations show that minimum‐bias error can be achieved by using a small bandwidth in estimating the spectra and by locating the microphones close to the sample. Furthermore, random error can be minimized by maintaining a high coherence between microphone signals. This implies that the microphones should have a small spacing. However, high coherence may not be realized when a microphone location coincides with a node point in the sound field.

62 citations


PatentDOI
TL;DR: In this paper, a blow detector senses the blowing action and triggers a bistable flip-flop which in turn drives a keying relay, and a speaking person blows a first time into a tube to key the processing equipment and blows a second time to unkey the equipment.
Abstract: In acoustical sound processing equipment, a combination microphone and blow actuated keying and unkeying means. The speaking person blows a first time into a tube to key the processing equipment and blows a second time to unkey the equipment. The person also vocalizes into the tube. The tube is in fluid communication with a chamber which houses the microphone. A blow detector senses the blowing action and triggers a bistable flip-flop which in turn drives a keying relay.

48 citations


PatentDOI
TL;DR: In this article, a conductive film, transparent to acoustic waves, positioned between the passage and the electret element, the film contacting a grounded part of the casing, can form an integral part of a microphone moisture barrier.
Abstract: An electret microphone has a metal shield around it in order to guard against electromagnetic interference. Acoustic waves reach the electret element through a passage in the shield. The electret response is undesirably affected by an electric field produced by body capacitance when the microphone is brought close to a user's mouth. To overcome this effect, the microphone has a conductive film, transparent to acoustic waves, positioned between the passage and the electret element, the film contacting a grounded part of the casing. The conductive film can form an integral part of a microphone moisture barrier.

48 citations


Journal ArticleDOI
TL;DR: In this article, a new technique for the absolute calibration of microphones is described, based on the measurement of acoustic particle velocity in a traveling-wave tube by means of a laser-Doppler system.
Abstract: A new technique for the absolute calibration of microphones is described. This is based on the measurement of acoustic particle velocity in a traveling‐wave tube by means of a laser‐Doppler system. From this measurement and a knowledge of the specific acoustic impedance of the air in the tube, the sound pressure is determined at a point in the tube where a microphone is mounted and, by measuring the output voltage of the microphone, its sensitivity is found. At 500 Hz, the method is accurate to within ±0.03 dB. No significant discrepancy was found between a calibration by this method and one by a precise reciprocity technique, although there is some residual uncertainty arising from the fact that one calibration is made in a small coupler and the other in an essentially free field. The result is seen as validating the assumption of reciprocity on which most precise microphone calibrations rely.

45 citations


Patent
12 Feb 1981
TL;DR: In this article, a differentiator is connected with a regulator which regulates a vent valve in such a manner that the pressure in the chamber during the measurement phase, in which the systolic and diastolic pressures are measured, reduces at a constant rate.
Abstract: There is disclosed blood pressure measuring equipment comprising a sleeve with an inflatable chamber and a microphone, and an appliance connected by a line with the sleeve. The appliance comprises a pressure sensor and a differentiator connected to the sensor output. The differentiator is connected with a regulator which in operation regulates a vent valve in such a manner that the pressure in the chamber during the measurement phase, in which the systolic and diastolic pressures are measured, reduces at a constant rate. A Korotkoff tone identifier, which is connected through further elements with the microphone and the differentiator, ensures that only those signals from the microphone which occur simultaneously with a heartbeat-induced pressure fluctuation are delivered as Korotkoff tones and evaluated for determination of the systolic and diastolic pressure.

45 citations


Journal ArticleDOI
TL;DR: In this paper, the phase and gain mismatch errors are corrected by measuring the transfer function between the two microphone systems exposing them to the same sound (phase and pressure levels) over a wide range of frequencies.
Abstract: The accuracy of measuring acoustic intensity using two closely spaced microphones is examined. The phase and gain mismatch errors are corrected by measuring the transfer function between the two microphone systems exposing them to the same sound (phase and pressure levels) over a wide range of frequencies. The accuracy of the measurement method was verified by creating a sound field in an anechoic room and by generating plane‐wave propagation inside a long length of pipe with an anechoic termination. The measurement accuracies were very satisfactory. This method has the advantage of eliminating the recording and processing of two sets of data required in the circuit switching technique.

43 citations


Patent
10 Aug 1981
TL;DR: In this paper, an annunciator/intercom enabled for operation by the remote control system is described. But the radio frequency signal is not considered. But it is assumed that the user can hear a telephone call when the television receiver is operated in the annunciators mode.
Abstract: A television receiver comprises a television system energized by the powerlines of a commercial power grid, a telephone network access system including a microphone disposed in close proximity with the audio speaker of the television system and a remote control system enabling the receiver to be operated in a television mode or a telephone mode. The television receiver according to the invention is characterized by having an annunciator/intercom enabled for operation by the remote control system. In the annunciator mode, the audio section of the television system is disabled and the microphone is connected through a radio-frequency signal transmitter to the power lines for generating a radio-frequency signal for local transmission. A frequency-modulator is included for modulating the radio frequency signal. At least one remotely located radio-frequency receiver coupled to the power lines is provided for receiving and demodulating the radio-frequency signal and rendering audible television receiver user voice signals. The television receiver user can therefore announce a telephone caller when the television receiver is operated in the annunciator mode. Alternately, intercom units may be employed to provide a two-way intercom conversation link between the television receiver user and a remotely located called party.

41 citations


PatentDOI
TL;DR: In this paper, an optical duct is connected to the sound transducer, which only enables an object, such as a display table or a data display unit, to be viewed completely in a specific position.
Abstract: PHD. 80-037 ABSTRACT In a device for adjusting a movable or electro-acoustic sound transducer an optical duct is connected the sound transducer, which only enables an object, such as for example a display table or a data display unit to be viewed completely in a specific position. This position is unambiguously reproducible and thus also the position of the sound transducer or microphone relative to the mouth of the speaker. The optical duct may be constituted by diaphragms or a tube, as the case may be with intermediate walls, or a phase grating or amplitude grating. In a special case, the object is a display device and the device in accordance with the invention further comprises a device for generating variable data on a display. Since during the speaker identification process the device for the gene-ration of variable data projects consecutive instructions on the display device, the speech recognition process can thus be controlled automatically. The speaker to be identified is then constantly forced to keep his mouth in the correct position relative to the microphone.

36 citations


PatentDOI
TL;DR: In this paper, a method of correcting the speech of a person affected by stammering comprises applying to the hearing organ of the person through an audio signal transmission channel his own speech delayed in time, in a frequency range the upper frequency limit of which is from 0.6 to 1.5 kilohertz.
Abstract: A method of correcting the speech of a person affected by stammering comprises applying to the hearing organ of the person through an audio signal transmission channel his own speech delayed in time. The application of the speech is carried out in a frequency range the upper frequency limit of which is from 0.6 to 1.5 kilohertz. The audio signal transmission channel is blocked during the pauses between the speech fragments pronounced by the person. A device for correcting speech comprises a microphone, a delay unit and an earphone, all connected in series, an adjusting element for adjusting the frequency spectrum of the signal transmitted from the microphone to the earphone, a switching circuit for interrupting the transmission of signal from the microphone to the earphone, and a threshold circuit having its input connected to the microphone and its output connected to the control input of the switching circuit.

Patent
Gregory Lese1, Donald H. Nash1
06 Apr 1981
TL;DR: In this paper, a portable frequency modulation transmitter (1) for voice or data is disclosed for operation at infrared frequencies, which comprises keyboard (200) or microphone (402) input capability and dual modes of use.
Abstract: A portable frequency modulation transmitter (1) for voice or data is disclosed for operation at infrared frequencies. The transmitter comprises keyboard (200) or microphone (402) input capability and dual modes of use. In the data mode, carrier is generated when the keyboard is operated and is not generated when the keyboard is idle in order to conserve power. In the voice mode, either voice or data may be transmitted, data having priority over voice. A series connection of two variable modulus counters (302, 303) and other counters (304, 305, 306) provide a frequency shift keyed data signal, marker data signal, marker data frames between data frames, parity insertion and other features. In the event that the local battery power level falls below a particular level, a particular word is inserted in the applied binary data input.

PatentDOI
TL;DR: In this article, a dual communication system including a microphone for mounting for use by a wearer of a protective face mask, the output signal from the microphone being communicated to both a speaker system for local, audible broadcast and to a transmitter for remote broadcast by radio signal.
Abstract: A dual communication system is disclosed including a microphone for mounting for use by a wearer of a protective face mask, the output signal from the microphone being communicated to both a speaker system for local, audible broadcast and to a transmitter for remote broadcast by radio signal.

Patent
11 Sep 1981
TL;DR: In this article, the authors proposed a method to positively receive an audio signal by controlling a microphone driver by an output signal from a sensor and directing the microphone in a voice generating direction.
Abstract: PURPOSE:To positively receive an audio signal by controlling a microphone driver by an output signal from a sensor and directing the microphone in a voice generating direction CONSTITUTION:The position of a user, that is, the audio signal generating direction is confirmed by a sensor 11 and its detection signal is inputted in an arithmetic circuit 8 Therefore, the arithmetic circuit 8 confirms the direction which the microphone 10 directs In a case where it differs from the position of the user (a), the arithmetic circuit 8 sends a drive signal to a microphone driver 12, and directs the microphone 10 to the user (a) Accordingly, the audio signal generated by the user (a) can be positively received by the microphone 10

PatentDOI
TL;DR: In this article, three microphones of first order sound pressure gradient unidirectional type are disposed in a casing having front and rear portions at both sides in such a manner that the center axes of the microphone units are aligned on the same axis.
Abstract: Three microphone units of first order sound pressure gradient unidirectional type are disposed in a casing having front and rear portions at both sides in such a manner that the center axes of the microphone units are aligned on the same axis. Two among the three microphone units are arranged to face the front portion of the casing, and remaining one microphone unit is arranged to face the rear portion of the casing. The output signals of the three microphone units are combined at a variable ratio so that the directivity of the microphone system can widely vary from nondirectivity via first order sound pressure gradient unidirectivity to second order sound pressure gradient unidirectivity. The output signals of two microphone units used for obtaining second order sound pressure gradient unidirectivity may be respectively applied to high pass filters before being combined, and the combined signal is fed to an equalizer. If an equalizer having modified frequency characteristic is employed, such high pass filters may be omitted.

Patent
23 Dec 1981
TL;DR: In this article, a noise canceling transmitter for voice communications comprising a casing having a principal surface opposed to the mouth of the user and three side surfaces facing upwardly, laterally and downwardly when the principal surface is so opposed.
Abstract: Noise canceling transmitter for voice communications comprising a casing having a principal surface opposed to the mouth of the user and three side surfaces facing upwardly, laterally and downwardly when the principal surface is so opposed. Noise canceling openings in the three side surfaces communicate noise to the back of a diaphragm in the transmitter microphone. Openings in the principal surface communicate both noise and the speaker's voice to the front of the diaphragm. The noise acts on both sides of the diaphragm and is thus canceled, while the voice acts only on one side of the diaphragm and vibrates it.

Journal ArticleDOI
TL;DR: In this paper, two spaced secondary sources in a duct energized in antiphase, with the microphone situated centrally between them, are used to cancel the noise from a source by the addition of further noise.

PatentDOI
TL;DR: In this paper, an amplitude-doppler circuit is utilized to predict the time of closest approach to a munition by a target by measuring the time interval between zero crossings of the second and third derivatives of the received acoustic wave amplitude function.
Abstract: An acoustic target sensor and ranging system automatically detects military targets and provides a munition firing signal at the appropriate target position and time. An amplitude-doppler circuit is utilized to predict the time of closest approach to a munition by a target by measuring the time interval between zero crossings of the second and third derivatives of the received acoustic wave amplitude function. The circuit produces a firing signal by logically ANDing the closest point of approach signal it develops with a signal that indicates when target range is within specified limits. The circuit is realized by means of conventional electronic zero crossing detectors, an up-down counter, dividers, sample and hold devices and voltage comparators. The acoustic target signal is obtained from an omni-directional microphone the output of which is amplified and rectified.

Patent
26 Jan 1981
TL;DR: In this paper, the ultrasonic system for detecting and distinguishing between superimposed and single sheets moving along a transport path, the frequency of an ultrasonic source is kept tuned to the natural frequency of the total system, so that the phase displacement detected when a sheet-like object is present between the source and ultrasonic receiver represents characteristics of the object itself.
Abstract: In an ultrasonic system for detecting and distinguishing between superimposed and single sheets moving along a transport path, the frequency of the ultrasonic source is kept tuned to the natural frequency of the total system so that the phase displacement detected when a sheet-like object is present between the source and the ultrasonic receiver represents characteristics of the object itself. When no sheet-like object is present the receiver and the source are connected in a feedback circuit producing a certain natural frequency comparable to that which occurs with proximities between a microphone and a loud speaker, and the frequency of an oscillator is tuned to this natural frequency. When a sheet-like object is present the tuned oscillator frequency is applied to the ultrasonic source and its phase is compared with the phase of the ultrasonic signal at the receiver.

Patent
18 Feb 1981
TL;DR: In this paper, a range expander (20a, 20b, 20c) is activated based on ambient noise level conditions as sensed, for example, by a microphone (12) which determines the ambient noise levels.
Abstract: To provide for response of audio reproduction equipment with the full dynamic level range, for example of music, intended by a composer, although radio signals transmitted have been compressed to reduce the signal-noise level, an expander circuit (20a, 20b, 20c) is activated based on ambient noise level conditions as sensed, for example, by a microphone (12) which determines the ambient noise level. If the ambient noise level, for example within certain frequency bands, as selected by filters (32a, 32b, 32c) is low, so that the full dynamic range (difference between low volume and high volume passages) can be reproduced, a range expander (20a, 20b, 20c) is controlled, for example within selected frequency bands, to reproduce the respective frequencies, or the overall signal, with a dynamic range greater than that of the received compressed signal and, preferably, corresponding to the dynamic range of the original signal which was then compressed for transmission. To prevent a microphone from responding not only to ambient noise but also to the reproduced program, the reproduced program level can be sensed, for example, by an amplitude detector (14a) and, when the reproduced program level is low, briefly entirely interrupt the program level while activating the microphone to sense then ambient noise level and, if the sensed ambient noise level is low, thus permit control of the dynamic range expansion circuitry to expand the dynamic range of the reproduced signal to improve listening conditions.

PatentDOI
TL;DR: A helmet with a built-in modulator and amplifier for voice alteration and projection is described in this paper, which includes a forward projecting nose housing having a viewing shield and an array of light emitting diodes.
Abstract: A helmet having a built-in modulator and amplifier for voice alteration and projection is disclosed. The helmet includes a forwardly projecting nose housing having a viewing shield and an array of light emitting diodes. The light emitting diodes are coupled to the amplifier through a driver in which selected groups of the diodes are illuminated according to the intensity of the audio signal delivered by the amplifier. A loud speaker is mounted within the nose housing and is enclosed by a sidewall baffle. A microphone is mounted on the outside of the baffle and is connected to an audio alteration unit within the enclosure which includes an analog multiplier, a sine wave oscillator and an audio amplifier. The multiplier has an audio output corresponding to the product of the voice signal produced by the microphone and the sine wave oscillator signal. This produces a modulation effect which simulates an alien sound.

Patent
21 Aug 1981
TL;DR: In this article, a two-way voice communication instrument with a microphone and a transducer is described, which includes a transmit-receiver circuit system comprising a transmit conditioning circuit and a receiver conditioning circuit.
Abstract: A voice communication instrument system for two-way voice communication is disclosed. The system includes a voice communication instrument having a microphone and a receiver transducer, and a transmit-receiver circuit system comprising a transmit conditioning circuit and a receiver conditioning circuit. The transmit conditioning circuit provides amplification and frequency response correction of microphone electrical voice signals. The receiver conditioning circuit provides linear compression limiting, amplification, and, if desired, frequency response correction to an incoming electrical voice signal prior to introduction to the receiver transducer. The receiver conditioning circuit is powered from the transmit circuit D.C. output voltage, at low voltage and low current. The receiver conditioning circuit has a transformerless input circuit, and further includes bias current control to the various circuit stages to eliminate the effects of supply voltage fluctuations.

PatentDOI
TL;DR: In this paper, an electret-type microphone element is mounted in a cylindrical-shaped microphone case having a central wall, and a ceramic piezoelectric element is secured against the wall with a weight pressing against the opposite side of the piezelectric elements.
Abstract: A microphone unit particularly adapted to be used as a built-in microphone for a device such as a tape recorder in which vibration-induced components in the output of the microphone are substantially eliminated. An electret-type microphone element is mounted in a cylindrical-shaped microphone case having a central wall. A ceramic piezoelectric element is secured against the wall with a weight pressing against the opposite side of the piezoelectric element. The outputs of the microphone element and the piezoelectric vibration pickup unit are combined in a circuit in which the characteristics of the output of the piezoelectric element are corrected so that substantially all of the vibration-induced component from the microphone element is cancelled.

Patent
21 Sep 1981
TL;DR: In this paper, a television/telephone loudspeaker system is described where feedback oscillation caused by having a speaker phone system simultaneously in the transmit and receive mode is minimized by microprocessor-controlled dynamic balancing of a bridge network coupled to a telephone isolation transformer.
Abstract: A television/telephone loudspeaker system is disclosed wherein feedback oscillation caused by having a speaker phone system simultaneously in the transmit and receive mode is minimized. Improved separation of transmit and receive signals is accomplished by microprocessor-controlled dynamic balancing of a bridge network coupled to a telephone isolation transformer to which the incoming and outgoing calls are provided. In addition, by microprocessor sampling of the television receiver's loudspeaker drive voltage from the telephone input line, dynamic proportional attenuation of microphone gain (AGC) is digitally accomplished in simulating full duplex operation. Bridge balancing is accomplished by means of test signals output by the microprocessor during quiet periods with various other audio signal refinements, such as room background noise cancellation, available with the constant monitoring of the microphone signal. Audio and visual indications of an incoming telephone call are provided by the television receiver's tuning system microcomputer which detects and decodes user remote commands and activates a phone relay and phone light emitting diode (LED) drivers.

Patent
15 Jun 1981
TL;DR: In this paper, a removably mounted tray is shown to be clampable to a microphone stand or the like, which includes two pivotally attached half sections, each of which includes an indentation for receiving and engaging the stand and the cord from a microphone.
Abstract: A removably mounted tray is clampable to a microphone stand or the like. The tray includes two pivotally attached half sections, each of which includes an indentation for receiving and engaging the stand and the cord from a microphone. A latch pivotally attached to one of the half sections engages a depending flange from each half section to mate the two half sections and retain the tray clamped to the stand.

PatentDOI
Jr. Robert Lee Wallace1
TL;DR: In this article, the coupling paths are arranged in pairs, such that for every element below a center line (102) there is an element above the line (111) and the relationship between the element pairs is nonlinear.
Abstract: Highly directional response patterns can be obtained by connecting microphones or loudspeakers with tubular coupling path structures. The coupling paths comprise a plurality of elements (110,111 . . . 157) arranged in pairs (110,111; 112,113; . . . 156,157) so that for every element (110) below a center line (102) there is an element (111) above the line. Furthermore, the relationship between the element pairs is nonlinear. The desired directional response comprises one main lobe and a plurality of substantially smaller lobes below a determinable threshold value. The elements may be a bundle of tubes (90) or a plurality of apertures (110,111, . . . 157) in a single tube (100).

Patent
18 Dec 1981
TL;DR: A sound responsive variable visual display (light organ) including an array 25 of light-emitting diodes arranged along a pair of orthogonal axes is described in this article.
Abstract: A sound responsive variable visual display (light organ) including an array 25 of light-emitting diodes arranged along a pair of orthogonal axes. A pair of counters (58, 60) having decoded outputs (30, 35) forming common connection points of the anodes and cathodes, each connection point corresponding to one point on one of the orthogonal axes, to activate one diode at a time. The counters are driven by independent voltage controlled oscillators (49, 50) with independent quiescent frequency adjustments (51, 52). The control voltage (38) is an electrical analog of an audio sound field in which the device is placed as detected by a microphone (12). Also shown are alternate arrangements for reversing the direction of indexing of the elements in the array along the axes, one to reverse counter direction in response to reaching predetermined high and low counts (71) and an alternate apparatus (48, 79) including a voltage controlled oscillator (48) for switching between an up and down counting direction.

PatentDOI
TL;DR: In this paper, a transformerless balanced connection to a microphone transmission line, either at the transmission end or at the reception end thereof, is proposed, where power is applied in a phantom connection by superimposing a DC voltage on two balanced signal conductors of the transmission line and using the shield thereof as a ground return line.
Abstract: A microphone output transmission circuit for a capactive microphone such as an electret microphone employs a differential amplifier to provide a transformerless balanced connection to a microphone transmission line, either at the transmission end or at the reception end thereof. Power is applied in a phantom connection by superimposing a DC voltage on two balanced signal conductors of the transmission line, and using the shield thereof as a ground return line. A pair of microphones with oppositely-directed sound-gathering planes can be connected to respective inputs of a differential amplifier at the transmission end, so that the two microphones together have a bidirectional characteristic.

Proceedings ArticleDOI
01 Oct 1981
TL;DR: In this article, a near field microphone array and a single sensor wedge shaped hot-film probe were used to investigate the location of shock noise production in unheated supersonic jets and demonstrate the existence of a large turbulent structure that collectively interacts and phases the motion of the downstream shocks.
Abstract: Shock noise associated with unheated supersonic jets were 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 1.5 and 2.0 convergent-divergent nozzles. Correlation measurements through each shock cell of a single underexpanded case with the Mach 1.5 nozzle were obtained between the hot-film probe and microphone array. The results show for the Mach number cases selected that the probe's response is primarily sensitive to velocity. 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, although they demonstrate the existence of some large perhaps turbulent structure that collectively interacts and phases the motion of the downstream shocks. The near field microphone correlations demonstrate that downstream shocks dominate shock noise production, and suggests the existence of a Doppler effect in near field of the sources. In addition broadband shock noise is found to also propagate at small angles to the jet axis.

PatentDOI
TL;DR: In this article, a method for detecting Korotkoff sounds using a pair of microphones carried by an inflatable cuff adapted to be wrapped around a limb of a patient whose blood pressure is to be determined.
Abstract: Apparatus and a method for detecting Korotkoff sounds using a pair of microphones carried by an inflatable cuff adapted to be wrapped around a limb of a patient whose blood pressure is to be determined. A Korotkoff sound is first sensed by one of the microphones to produce a first signal which is coupled to a delay unit which has an output signal delayed by a certain time interval, such as 12 millisecond. At the end of this interval, a window is opened. The same Korotkoff sound is later sensed by the second microphone whose output signal is applied to the window and passes through the window so long as it is open. The output signal then actuates a recorder, such as an oscillograph.