Showing papers on "Microphone published in 1988"

A microphone array with adaptive post-filtering for noise reduction in reverberant rooms

Abstract: The author presents a self-adapting noise reduction system which is based on a four-microphone array combined with an adaptive postfiltering scheme. Noise reduction is achieved by utilizing the directivity gain of the array and by reducing the residual noise through postfiltering of the received microphone signals. The postfiltering scheme depends on a Wiener filter estimating the desired speech signal and is computed from short-term measurements of the autocorrelation and cross-correlation functions of the microphone signals. The noise reduction system has been tested experimentally in a typical office room. The system produces an enhanced speech signal with barely noticeable residual noise if the input SNR is greater than 0 dB. The received noise power-measured in the absence of the speech signal-can be reduced by 28 dB. >

Adaptive, programmable signal processing and filtering for hearing aids

Abstract: A hearing aid system utilizes digital signal processing to correct for the hearing deficit of a particular user and to maximize the intelligibility of the desired audio signal relative to noise. An analog signal from a microphone is converted to digital data which is operated on by a digital signal processor, with the output of the digital signal processor being converted back to an analog signal which is amplified and provided to the user. The digital signal processor includes a time varying spectral filter having filter coefficients which can be varied on a quasi-real time basis to spectrally shape the signal to match the hearing deficit of the user and to accommodate ambient signal and noise levels. The coefficients of the spectral filter are determined by estimating the energy in several frequency bands within the frequency range of the input signal, and using those energy estimates to calculate desired gains for the frequency bands and corresponding spectral filter coefficients. The spectral energy analysis may be carried out using pairs of high pass and low pass digital filters in cascade relation, with the output of each low pass filter being provided to the next pair of high pass and low pass filters. The rate at which output data is provided from the filters in each pair may be reduced from the sample rate of input data by one half for succeeding pairs of filters in the cascade to thereby reduce the computation time required.

Adaptive, programmable signal processing hearing aid

Abstract: A hearing aid system utilizing digital signal processing is programmable to fit the hearing deficit of a particular user and adaptive to the sound environment to maximize the intelligibility of the desired audio signal relative to noise. A signal picked up from microphone (30) is amplified and filtered by preemphasis circuit (32) and converted to digital data through an 8-bit linear A-to-D converter (47). The processor (50) preferably performs spectral shaping on the data to match the user's preference and performs a non-linear adaptive amplification function on the digital data. The amplification gain function may include several piecewise linear sections, including a first section providing expansion up to a first knee (K1), a second section providing linear amplification from the first to a second knee (K2), and a third section providing compression for signals above the second knee to reduce the effect of over range signals and minimize loudness discomfort to the user.

Device for non-invasive diagnosis and monitoring of articular and periarticular pathology

Abstract: Arthritis and related diseases are diagnosed by recording sound from a joint moving against a predefined resistance or a controlled velocity, and correlating the acoustic amplitude and instantaneous frequency with the joint angle. A preferred apparatus includes an articulated restraint for receiving a limb, a hydraulic pump and flow control valve for resisting joint movement, a pressure gage for measuring the torque about the joint, a position sensor sensing the joint angle, a noise cancelling microphone or an accelerometer for sensing sound from the joint, an amplifier and filter for amplifying and filtering the acoustic signal from the microphone or accelerometer, a frequency-to-voltage converter and an amplitude detector for determining the instantaneous frequency and amplitude of the filtered acoustic signal, an analog-to-digital converter for obtaining simultaneous numerical samples of the instantaneous frequency, amplitude, and torque, and a microcomputer programmed to display and record the instantaneous frequency and amplitude as a function of the joint angle, measured by either a digital shaft encoder or a potentiometer.

Apparatus and method for automatically monitoring broadcast band listening habits

Abstract: An electronic monitor and system for automatically monitoring broadcast band listening habits and periodically collecting data at a central location. The electronic monitor includes a microphone circuit and a tuner circuit. The tuner circuit is tuned until a match is found between the microphone circuit and the tuner circuit and the event of a match is recorded in the portable monitor. Operation may be performed intermittently to extend battery operation. The portable monitor further has power supply monitoring circuits to prevent incorrect data and loss of data in the event of battery discharge. Digitizing and signal processing circuits are provided to perform the match and loss of match determination.

Digital sound data storing device

Abstract: This invention relates to a digital sound data storing device wherein analog sound data is synthesized from digital sound data. The device includes a case and an IC card detachably attached to the case. A plurality of digitized sound data groups are stored at a built-in memory of the IC card. The digitized sound data groups are obtained by recording sentences for foreign language conversation practice through a microphone to thereby obtain analog sound signal which is converted to digital sound signal. When the IC card is attached to the case, a signal processing circuit operates to read out the digital sound data group, which is converted to the analog sound signal. The analog sound signal is supplied to an amplifying circuit and reproduced as sound.

Headset having a post auricle mount and arranged to be worn on a person's outer ear

Abstract: A headset (1) with sound receiving microphone (6) and sound reproducing transducer (3) is arranged to be worn on a person's outer ear (2). For this purpose, the headset has a post auricle mount (9) with a lengthwise adjustable stabilizer (10) behind the ear, said post auricle mount being secured to the sound reproducing transducer housing (3) in a pivotable manner, hereby enabling it to be swivelled in several directions.

Noise reduction system

Abstract: In order to overcome problems in an active noise reduction system of sound buffets at low frequency and signal enhancement caused by imperfect transfer functions of a noise cancelling sound generator and a microphone, one or more high pass filters for reducing low frequency signals are provided in a feedback loop between the sound generator and microphone. A low pass filter is provided for extending the bandwidth of the system but which does not introduce unduly large phase shifts.

Vehicular hands-free telephone system

Abstract: A voice of user is picked up by a microphone built in a sun visor of a vehicle. The output of microphone is frequency-modulated by an FM transmitter powered by a solar battery and transmitted through a transmitting antenna in a vehicle. The frequency-modulated radio signal is received and demodulated by a receiver with a receiving antenna. The received signal is further transmitted outside of the vehicle through a transceiver.

Head diffraction compensated stereo system

Abstract: A stereo audio processing system for a stereo audio signal processing reproduction that provides improved source imaging and simulation of desired listening environment acoustics while retaining relative independence of listener movement. The system first utilizes a synthetic or artificial head microphone pickup and utilizes the results as inputs to a cross-talk cancellation and naturalization compensation circuit utilizing minimum phase filter circuits to adapt the head diffraction compensated signals for use as loudspeaker signals. The system provides for head diffraction compensation including cross-coupling while permitting listener movement by limiting the cross-talk cancellation and diffraction compensation to frequencies substantially below approximately ten kilohertz.

Head diffraction compensated stereo system with optimal equalization

Abstract: A stereo audio processing system for a stereo audio signal processing reproduction that provides improved source imaging and simulation of desired listening environment accoustics while retaining relative independence of listener movement. The system first utilizes a synthetic or artificial head microphone pickup and utilizes the results as inputs to an equalization circuit with the outputs coupled to a cross-talk cancellation compensation circuit utilizing minimum phase filter circuits to adapt the head diffraction compensated signals for use as loudspeaker signals. The system provides for head diffraction compensation including equalization and cross-coupling while permitting listener movement by modifying the cross-talk cancellation and diffraction compensation at frequencies substantially above approximately ten kilohertz while maintaining substantially accurate equalization for the desired incidence angle.

Air buoyant piezoelectric polymeric film microphone

Abstract: Piezoelectric polymer film, when conformably adhered to inner or outer surfaces of an inflated balloon, functions as a microphone when the pressure of sound waves causes the film to vibrate and generate an electrical signal. The film may be in the form of an helical strip, individual strips electrically serially connected, or may itself form the inflatable material. A receiving device is electrically coupled with the electrodes formed on the piezoelectric film for processing the electrical signals generated by the vibrating piezoelectric polymer film.

Sound pattern discrimination system

Abstract: A system is provided for the detection and recognition of pre-established sound patterns such as the various patterns produced by the sirens of emergency vehicles. The system includes a microprocessor which programmed with pre-established sequence detection algorithms corresponding to the different types of emergency vehicles sirens signal patterns which are to be recognized. A first omnidirectional microphone is coupled through a bandpass circuit to a trigger circuit to produce square wave signals representative of analog signals in the band of interest. At least two directional microphones are coupled through similar bandpass amplifier circuits to analog digital converters which produce a digital output representative of the strength of the signals received by the directional microphones. This directional information along with the output of the Schmitt trigger is supplied to the microprocessor which is used to control the signal lights at an intersection in response to the detected siren.

Active acoustic attenuation system with differential filtering

Abstract: An adaptive active acoustic attenuation system is provided with extended frequency range to attenuate undesired noise which was previously filtered out to avoid instability of the adaptive model. The input signal from the input microphone to the model and the error signal from the error microphone to the model are differentially bandpass filtered to provide a narrower frequency range error signal. In one embodiment, the model operates in its stable region to provide accurate well behaved correction signals to the cancelling loudspeaker, while still receiving a low frequency input noise signal from the input microphone including frequencies below such range. Minimum attenuation frequency has been reduced by at least an octave.

Impulse excitation technique for dynamic flexural measurements at moderate temperature

Abstract: The impulse excitation technique (IET), which is presently a precise and reliable technique for measuring dynamic moduli at room temperature, has been adapted to measure dynamic flexural modulus at temperatures in the range of 25° to 300 °C. This modified technique involves a sensitive microphone and electronics to record and analyze the sound waves emitted from a specimen vibrating in the fundamental flexural mode. The fundamental resonant frequency and geometry of the specimen are used to obtain the modulus. The location of the microphone relative to the specimen is critical and is a major factor once the specimen is placed within the heated environment. Problems were identified and solved, and test data for aluminum are presented to support the modification of the IET for use at elevated temperatures.

Telephone receiver transmitter device

Abstract: A telephone speaker/microphone device for suspending in a user's ear to enable hands-free communication without common occurrence of feedback. The device utilizes control circuitry capable of interconnecting microphone and speaker leads from a telephone or other communicating device to a speaker element and microphone element which are embodied in a single earpiece and suspended at the user's ear. The speaker element and microphone element are positioned sufficiently close and are wired such that the speaker signal is approximately 180 degrees out of phase with the microphone signal and results in cancellation of feedback by virtue of the out of phase relationship. This device and method are particularly useful in communication devices which utilize a side tone electronically transmitted from the microphone into the speaker element at the user's ear. Such side tone signal can be utilized in an out-of-phase relationship to cancel ambient feed back which has historically prevented close proximate use of microphone and speaker elements.

Probe-tube microphone measures in hearing-impaired children and adults

Abstract: This study was designed to investigate the reliability of real-ear measurements of sound pressure level (SPL) and to compare these values with two coupler measures of SPL. A commercially available probe tube microphone system was used to measure real ear SPL in both children and adults. Test-retest reliability decreased as a function of frequency for both groups and, in general, was slightly poorer for the children. For both groups, coupler to real ear differences were larger for the 2 cm3 coupler than for the reduced volume coupler; however, no significant differences were observed between groups. In addition, a measure of ear canal volume was not found to be a good predictor of coupler to real ear discrepancies.

Transceiver for hanging on an ear

Abstract: Several embodiments of transceivers including an earphone and a microphone carried by a mounting device that is adapted to be mounted on the ear of the user. The mounting device resiliently biases the earphone into engagement with the auditory canal of the user so as to improve sound transmission. A number of arrangements of mounting of the microphone and shielding from noise are shown and described.

Method and system of background noise suppression in an audio circuit particularly for car radios

Abstract: To improve the fidelity of reproduction of automobile radios during conditions of extraneous in the passenger compartment, noise signals are derived, for example from a microphone installed in the engine compartment of the vehicle The audio program signals are conducted through a multichannel equalizer; the program audio signals and the noise signals are separated into a plurality of frequency bands, for example three bands, the envelope of which are demodulated and the derived demodulation signals are instantaneously compared to then control the transfer characteristics of the respective bands of the equalizer in accordance with the derived comparison between the respective bands of the noise and program signals Preferably, amplitude, change of transfer amplitude and rate of change is controlled by weighting the comparison signals and the rate of change thereof in relation to the frequency bands and the extent of change of the signals

Noise suppression apparatus

Abstract: A noise suppression apparatus has a main microphone for mainly picking up a voice and for outputting an input signal including an audio signal and a first noise component generated from a noise source, a reference microphone for picking up a second noise component generated from the noise source, a filter bank for band-dividing the input signal from the main microphone and the second noise component from the reference microphone, and a noise cancel circuit for obtaining a phase difference between the input signal and the second noise component with respect to each divided band of the filter bank so as to correct the input signal based on the phase difference and for cancelling the first noise component in the input signal by use of the corrected input signal.

Voice control circuit for a communication terminal

Abstract: A voice control circuit controls the volume range of the voice signals in a communication terminal, such as a video telephone having a hands-free facility. In a hands-free facility, the signal-to-noise ratio is lower than in telephones with handset. To avoid audible feedback, hands-free facility conventionally includes a volume control circuit which decreases or increases the gain in the receive path as the gain in the set path increases or decreases. The problem to be solved is to utilize the maximum possible loop gain without reducing the gain in the receive path. The idea is to place behind the microphone (50) a controllable compandor (52) in which the signal voltages from the microphone (50) are A. compressed if they lie above a predetermined level, B. expanded if they lie below the predetermined level, and C. imparted maximum gain if they have the predetermined level. The predetermined signal level, which corresponds to the compression set point, is varied by a control circuit (90) as a function of the ambient noise.

Transmission system for telephone communications or data transfer

Abstract: The invention concerns a transmission system for telephone communications comprising at least two telephone sets, each incorporating a microphone (9) and a receiver and/or loudspeaker, a tele­phone line connecting the telephone sets by at least two conductors in order to form a current loop. Connected to the current loop is an alternating-current generator (2) and current-­steering switch elements (11), arranged to steer the current to be routed during the positive half-cycles generated by the generator via the microphone circuit (9) of the first set and the receiver and/or loudspeaker circuit (10′) of the second set as well as to steer the current during the negative half-cycles via the receiver and/or loudspeaker circuit (10′) of the first set and the microphone circuit (9) of the second set, whereby voice transmission takes place by sampling the microphone signals of the inter-­connected telephone sets and transmitting the alternate signal samples superimposed on the generator signal so that the microphone signal of the first set is superimposed on the positive half-cycles of the generator signal while the microphone signal of the second set is super­imposed on the negative half-cycles of the gener­ator signal. The invention is also applicable to data transmission, wherein the telephone sets are replaced by modems of a simple construction.

Hands-free type automobile telephone and general-purpose telephone

Abstract: The automobile telephone is used in a hands-free condition by coupling its handset with a cradle-shaped coupler which is provided with a pair of swingable support plates for receiving a transmitter portion and a receiver portion of the handset in use to make it possible to receive the handset regardless its size and mounting angle. On a back surface of the swingable support plates are provided a speaker and a microphone which are coupled with the transmitter portion and the receiver portion of the handset respectively so as to make it possible to use the telephone in a hands-free condition through these speaker and microphone incorporated in the cradle-shaped coupler.

Conversation device of MR imaging apparatus

Abstract: The conversation device of an MR imaging apparatus including a magnet for generating a static magnetic field, a gradient field coil disposed between pole pieces of the magnet and used for endowing an NMR signal with positional information, a high-frequency coil disposed in the gradient field coil for transmitting an electromagnetic wave and for detecting the NMR signal, and a bed disposed in the high-frequency coil for holding a person to be inspected, is disclosed. In this conversation device, a bone conduction microphone insensitive to a magnetic field and surrounding noise is mounted on an auxiliary member (which is used for placing the to-be-inspected person at a predetermined position on the bed) so that the bone conduction microphone can be kept in contact with that part of the to-be-inspected person which can conduct the voice of the to-be-inspected person by a bone.

Moving animal toy

Abstract: A moving animal toy wherein when a motor is started upper and lower beaks are pivoted up and down simultaneously to widely open and close its beaks, and at the same time a head is slowly rotated twisting its head sideways. When a voice is uttered against the toy, it is received through a microphone by a voice recording and reproducing device and after a specified recording time elapses, the voice is reproduced by the speaker.

An analytical, one‐dimensional, standing‐wave model of a linear active noise control system in a duct

Abstract: Assuming plane waves, and making use of the transfer matrices and electroacoustic analogies, a standing wave analysis of the active noise control system in a duct is presented incorporating the characteristics of the primary source as well as the auxiliary source. Analytical expressions have thus been derived for the ratio of the two source pressures and certain other ratios or relations of interest for complete or partial cancellation of noise downstream of the auxiliary source. The resulting closed‐form expression for the filter transfer function H0 reveals the real role of the auxiliary source both when it is on and off, and the importance of the distance between the input microphone and the auxiliary source (generally, a loudspeaker). Finally, the analysis is extended to incorporate the convective and dissipative effects of the meanflow and viscothermal damping.

Method and apparatus for improving voice intelligibility in high noise environments

Abstract: The output of a noise canceling first or second order electret gradient microphone in a very high noise environment has background noise attenuated and certain phonemes shaped to improve intelligibility by means of a slow action automatic gain control circuit (22) which matches dynamic range of the microphone (10) with dynamic range of the following communication channel, and a fast action automatic gain control circuit (24) in parallel with the first circuit, that reshapes puff noise into plosive, fricative and affricative speech component signals that produced the puff noise.