Showing papers on "Microphone published in 1986"

Influence of errors on the two‐microphone method for measuring acoustic properties in ducts

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

Microphone actuation control system suitable for teleconference systems

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

A new application of adaptive noise cancellation

Abstract: A new application of Widrow's adaptive noise cancellation (ANC) is presented in this paper. Specifically, the method is applied to the case where an acoustic barrier exists between the primary and reference microphones. By updating the coefficients of the noise estimation filter only during silence, it is shown that ANC can provide substantial noise reduction with little speech distortion even when the acoustic barrier provides only moderate attenuation of acoustic signals. The use of the modified ANC method is evaluated using an oxygen facemask worn by fighter aircraft pilots. Experiments demonstrate that if a noise field is created using a single source, 11 dB signal-to-noise ratio improvements can be achieved by attaching a reference microphone to the exterior of the facemask. The length of the ANC filter required for this particular environment is only 50 points.

High fidelity earphone and hearing aid

Abstract: A high fidelity earphone or hearing aid utilizes an acoustic path from a location near where the sound is delivered to the ear, to a location near the backside of the sound-producing diaphragm. A vent to the atmosphere from a location near the backside of the sound-producing diaphragm is also taught. A microphone on the earphone makes it safe to listen to the radio or a tape player in public, because of the capability of hearing outside sounds.

Methods of measuring the attenuation of hearing protection devices

Abstract: The published literature describing three real-ear-attenuation-at-threshold (REAT), nine above-threshold, and four objective methods of measuring hearing protector attenuation is reviewed and analyzed with regard to the accuracy, practicality, and applicability of the various techniques. The analysis indicates that the REAT method is one of the most accurate available techniques since it assesses all of the sound paths to the occluded ear and, depending upon the experimenter's intention, can reflect actual in-use attenuation as well. An artifact in the REAT paradigm is that masking in the occluded ear due to physiological noise can spuriously increase low-frequency (less than or equal to 500 Hz) attenuation, although the error never exceeds approximately 5 dB, regardless of the device, except below 125 Hz. Since the preponderance of available data indicates that attenuation is independent of sound level for intentionally linear protectors, the use of above-threshold procedures to evaluate attenuation is not a necessity. An exception exists in the case of impulsive noises, for which the existing data are not unequivocal with regard to hearing protector response characteristics. Two of the objective methods (acoustical test fixture and microphone in real ear) are considerable time savers. All objective procedures are lacking in their ability to accurately determine the importance of the flanking bone-conduction paths, although some authors have incorporated this feature as a post-measurement correction. The microphone in real-ear approach is suggested to be one of the most promising for future standardization efforts and research purposes, and the acoustical test fixture technique is recommended (with certain reservations) for quality control and buyer acceptance testing.

Hearing aid with locating microphones

Abstract: A hearing aid which can be built into a frame, such as eyeglasses, to be worn by a hearing-impaired person has a first microphone arrangement having a directional reception pattern, and a second microphone arrangement for sound locating. The second microphone arrangement includes a first locating microphone disposed in the region of one ear of the hearing-impaired person, and a second locating microphone disposed in the region of the other ear. The signal from both locating microphones are mixed by a low-pass filter with the signal from the microphone arrangement having a directional reception pattern, and the output of the mixing operation is supplied in common to both ears of the hearing-impaired person.

Speakerphone for radio and landline telephones

Abstract: An improved speakerphone (120 and 130 in FIG. 1) for radio and landline telephones is described. The improved speakerphone (120 and 130) includes a microphone (102 and 132), a speaker (104 and 134) and unique control circuitry (106 and 136). The control circuitry of speakerphone (200 in FIG. 2) interfaces a microphone (250) to a transmit signal (220) and speaker (260) to a receive signal (222) of a duplex communication path, such as a radio channel or telephone line. An audio switch (212) opens or closes the speaker audio path in response to a control signal (224) from control logic (230), and another audio switch (202) opens or closes the microphone audio path in response to the binary complement of the control signal (224). Transmit and receive signal detectors (206 and 207), each includes a logarithmic amplifier (240), an envelope detector (241), a smoothing filter (245 ), a valley detector (242), a summer (243) and a comparator (244) for detecting the presence of audio signals in environments that may be subject to high background noise. Binary output signals from the transmit and receive signal detectors (206 and 207) are applied to control logic (230) which generates the control signal (224) for opening and closing the transmit and receive audio paths. The control logic (230 in FIG. 3) includes delay circuitry (316 and 318) and logic circuitry (304, 306, 308, 310, 312 and 314) for setting and resetting a flip-flop (302) storing the control signal (224). The control logic (230) changes the state of the control signal flip-flop (302) for switching the audio path between the micorphone and speaker when audio signals from the presently closed audio path have not been detected for a time interval determined by the delay circuitry (316 and 318) and audio signals thereafter are detected on the other audio path.

Device for preventing of snoring

Abstract: A device for preventing snoring of a sleeping person comprises an arrangement (5-9) for detecting snoring sounds and an apparatus (17) controlled by this arrangement and adapted to influence the person to stop snoring. The arrangement has a sound-receiving microphone (5) and at least one frequency filter (6, 8), arranged to deliver signals deriving from sounds with frequencies typical for snoring sounds in order to determine if the present sound derives from snores. The arrangement further comprises a circuit (10-14) for determining if the signals delivered by the microphone are periodically appearing at time intervals, which are typical for snores. The circuit comprises a counter (13) which is arranged to count the number of snores. The arrangement (5-14) of the device is adapted to send activating control pulses to the apparatus when the counter (13) has detected a predetermined number of successive snores with the time intervals.

Directional-microphone arrangement

Abstract: A directional microphone arrangement has a number of discrete microphones each having an electrical output and elements for deriving first and second electrical signals which are 180° out of phase with respect to each other from the outputs of the microphones. A third signal is also derived through differentiation from one of the microphone outputs which is 90° out of phase with respect to one of the first or second signals. All of the first, second and third signals are added in a summing unit, the output of the summing unit forming the output for the arrangement.

Wireless phone system communicatively combined with a computer

Abstract: A wireless phone system comprises a master phone unit connectable to a telephoneline, and a wireless phone receiver connectable to the master phone unit, the wireless phone receiver comprising a computer circuit for processing data as a computer, a wireless circuit for wirelessly transmitting information including the computer data, microphone means for inputting sound information, speaker means for outputing the sound information, and switching means for selecting whether the microphone means and the speaker means are connected with the computer circuit, or not. The master phone unit has a computer circuit.

Transceiver battery charging apparatus and method

Abstract: An apparatus and method for recharging a rechargeable battery in a hand-held transceiver while maintaining communications capability through the transceiver. The battery charger is housed in a charging unit to which the transceiver can be removably attached. A dual-mode charging circuit generates a first voltage during transmit mode and a second, higher voltage during receive mode, and includes a DC-DC converter for generating the second voltage. The charger includes a push-to-talk (PTT) switch and mode control circuitry for simultaneously controlling the operating modes of the charger and transceiver, and additionally includes a separate microphone, speaker, and audio and RF amplifiers for providing greater RF power and audio power while the transceiver battery is being recharged.

Handbook of Recording Engineering

Abstract: Foundations in Acoustics.- Acoustics in the Modern Studio.- Psychoacoustics: How We Hear.- Microphones.- Microphones: Basic Principles.- Microphones: The Basic Pickup Patterns.- Environmental Effects and Departures from Ideal Performance.- Microphones: Electronic Performance and the Electrical Interface.- Microphone Accessories.- Recording Systems: Analysis, Architecture, and Monitoring.- Basic Audio Signal Analysis.- Recording Consoles, Metering, and Audio Transmission Systems.- Monitor Loudspeakers.- Recording Technology.- Analog Magnetic Recording and Time Code.- Digital Recording.- The Digital Postproduction Environment.- Signal Processing.- Equalizers and Equalization.- Dynamics Control.- Reverberation and Signal Delay.- Special Techniques in Signal Processing.- Recording Operations.- Fundamentals of Stereo Recording.- Studio Recording and Production Techniques.- Classical Recording and Production Techniques.- Surround Sound Recording Techniques.- Production Support Functions.- Mixing and Mastering Principles.- Music Editing and Assembly.- Consumer Media.- Recorded Tape Products for the Consumer.- Optical Media for the Consumer.- The Stereo Long-Playing (LP) Record.- Studio Design Fundamentals.- Recording Studio Design Fundamentals.

Directional microphone apparatus

Abstract: A directional microphone apparatus includes an array of at least three sets of microphone units, circuits for processing the output signals from the microphone units to produce a first signal which varies in accordance with the second order bidirectional sound pressure gradient characteristic of the array, weighting circuits for applying respective weighting coefficients to the output signals from the microphone units in accordance with a desired shape of directivity response characteristic, to produce corresponding weighted signals which are combined to form a second signal. The first and second signals are combined to produce an output signal which exhibits sharp directivity even at relatively low frequencies in the audio range.

Transfer function technique for impedance and absorption measurements in an impedance tube using a single microphone

Abstract: Using a periodic pseudorandom sequence as the noise source, it is possible to implement the two‐microphone transfer function technique for impedance and absorption measurement in an impedance tube with a single microphone, thereby eliminating the elaborate calibrating procedure and any error associated with phase‐mismatching. Results obtained by the proposed procedure compared well with those obtained by the standard standing‐wave‐ratio method.

Electronic sound amplifier stethoscope with visual heart beat and blood flow indicator

Abstract: An electronic sound amplification stethoscope including a battery powered self-contained sound amplification circuit contained in a hand held connector housing inserted in the flexible sound conduit of the stethoscope. The circuit includes a miniaturized microphone for receiving sound waves from the stethoscope pickup head and a miniaturized speaker for transmitting amplified sound waves to the stethoscope headpiece. Both the microphone and speaker may be housed in the connector housing and are vibration insulated from the housing itself. The electronic circuitry and battery power source are located in compartments separated from the microphone and speaker. A LED light source is placed in series between the amplifier circuit and the power source such that the fluctuations in its intensity are directly proportional to the power surges in the circuit. The LED is thus a visual indicator of such body functions as respiration and blood flow. The amplifier circuit is also provided with an on/off volume control thumbwheel located on the surface of the connector housing.

Method of using an opto-acoustic apparatus for measuring concentration of gas

Abstract: A method of using an opto-acoustic apparatus for measuring, in an environment containing extraneous noise, the concentration of a gas in a mixture of gases or of particulates in a gas. The method is constituted by the steps of providing a measuring opto-acoustic cell having gas inlet and gas outlet for receiving and discharging a gas containing a particulate or a mixture of gases containing a gas the concentration of which is to be measured, directing laser rays from a laser ray generating device into the opto-acoustic cell, placing a chopper in the path of the laser rays between the device and the cell and operating the chopper for chopping the laser rays at a frequency corresponding to the resonant frequency of the cell, providing a narrow band microphone having a resonator with a narrow resonance frequency range including the resonant frequency of the cell and sufficiently narrow to exclude unwanted noise signals from the environment in which the cell is located, placing the microphone on the cell for detecting the sound signal generated by changes in the internal pressure of the cell, and determining from the sound signal the concentration of the gas in the mixture of gases or the concentration of the particulates in the gas.

Adaptive optimization of microphone arrays under a nonlinear constraint

Abstract: In this paper we describe an algorithm for optimizing a microphone array with the objective of receiving a desired speech signal from a known direction, in the presence of broad band noise sources with unknown locations and possibly slowly varying characteristics. Each microphone feeds a tapped-delay-line filter, and the outputs of the filters are summed to give the final output. The tap weights of the filters are adjusted adaptively to minimize the output power subject to a constraint on the allowed distortion of the desired speech signal. Since the quality of a speech signal is quite insensitive to phase distortion, the constraint is on the magnitude transfer function from the source to the output. It is this choice of constraint that differentiates our approach from other methods of adapting arrays.

Extension of the two‐microphone transfer function method for impedance tube measurements

Abstract: The two‐microphone transfer function method for impedance tube measurements has been extended to include the effect of the tube attenuation.

Video camera microphone with zoom variable acoustic focus

Abstract: In a video camera having a zoom lens and a microphone pickup for recording sound associated with the video images, the acoustic focus (or acoustic acceptance angle) of the microphone pickup is varied between uni-directional and omni-directional in synchronism with the zoom lens control of the camera in order to correlate the optical field of view of the lens with the acoustic field of view of the microphone for optimum coordination of picture and sound.

Noise control characteristics of synchrophasing. II - Experimental investigation

Abstract: A simplified cylindrical model of an aircraft fuselage is used to investigate the mechanisms of interior noise suppression of the synchrophasing technique. This investigation allows isolation of important parameters to define the characteristics of synchrophasing. The optimum synchrophase angle for maximum noise reduction is found for several interior microphone positions with pure tone source conditions. Noise reductions of up to 30dB are shown for some microphone positions, however, overall reductions are less. A computer algorithm is developed to decompose the modal composition of the cylinder vibration over a wide range of synchrophase angles. The circumferential modal response of the shell vibration is shown to govern the transmission of sound into the cylinder rather than localized transmission.

Hearing aid with adjustable frequency response

Abstract: A hearing aid has a sound-receiving microphone from which an incoming signal is supplied to a number of different channels, each channel being allocated to a different frequency range within a total expected range of frequencies for the incoming signal. Each channel includes a circuit for measuring the strength of the signal within the frequency range for that channel and for changing the respective strengths of the signals in the other channels by suppressing weak signal channels in favor of strong signal channels.

Audible sound transmission system

Abstract: An audible sound transmission system is proposed which comprises at least two radio transmitting and receiving stations. A radio transmitting and receiving station exhibits a radio transmitting and receiving device (17), including a transmitting and receiving antenna (18) and a power source (19), which is accommodated in a housing which is normally intended for a hearing aid to be worn behind the ear or in the ear. The housing (16) is connected to a hearing aid clip (15) through which a connecting line (14) connecting the radio transmitting and receiving device to an electro-acoustic sound transducer (13) is conducted. The sound transducer, acting simultaneously as receiver and microphone, is located in an otoplasty (10) which contains a sound channel (11) connected to the sound output (12) of the sound transducer.

Programmable sound reproducing system

Abstract: In a hearing aid system, selected optimum parameter values are programmed into an electronically erasable, programmable read-only memory (EEPROM) (84) which supplies coefficients to a programmable filter (64) and amplitude limiter (67) in the hearing aid so as to cause the hearing aid to adjust automatically to the optimum set of parameter values for the speech level, room reverberation, and type of background noise then obtaining. The programmable filter may be a digital equivalent of a tapped delay line in which each delayed sample is multiplied by a weighting coefficient, and the sum of the weighted samples generates a desired electro-acoustic characteristic; or a tapped analog delay line in which the sum of the weighted outputs of the taps generates the desired characteristics. Acoustical feedback is reduced by an electrical feedback path in the hearing aid which is matched in both amplitude and phase to the acoustic feedback path, the two feedback signals being subtracted so as to cancel each other. Alternatively, a single filter in the forward path may be used for this purpose with a transmission characteristic equivalent to that of the programmable filter in the forward path plus the electrical feedback path. Also, the relative speech-noise content in the signals from the hearing aid microphone is sensed and binary words are generated and supplied to the programmable filter for selecting from memory a set of delay line tap coefficients that are effective to impart to the filter the appropriate frequency response for the specific environmental noise condition being detected.