Showing papers on "Microphone published in 1995"

Wireless and wired communications, command, control and sensing system for sound and/or data transmission and reception

Abstract: An interactive microprocessor based wireless communication device includes sound and data transceivers, signal detection and coupling devices, signal conversion device, voice recording, playback and storage device, voice activated device, display device, touch screen or similar device, sensors, frequency generation device, sound detection and reproduction devices and power source to concurrently perform generalized two way wireless communications, command, control and sensing functions utilizing radio and infra-red frequency communication links. A microprocessor receives signals from the touch screen and generates a digital data, command/or control signal for transmission to external devices such as home appliances and remote sensors. The microprocessor also responds to voice signal commands received via microphone and a voice processor. The microprocessor uses this signal to generate data, command/or control signals for transmission to external devices such as telephone, paging and intercom systems. Sound signals may be stored in a voice recorder and playback IC for subsequent message processing and coupling to a transceiver and/or a speaker. Telephone ringer signals are generated by the microprocessor and are coupled to a ringer for audio output. In response to certain commands, the wireless communication device establishes a communication link with external devices using radio frequency or infra-red frequency transmission and/or reception. Sensor signals are created by sensors that can detect physical differential changes and that can convert the changes into measurements. These signals are coupled to the microprocessor for further processing, display and/or transmission.

Noise cancellation apparatus

Abstract: Apparatus for reducing acoustic background noise for use with a telephone handset (10) or a boom microphone device (100) or a boom headset (401) or the like. The apparatus includes first (12) and second (14) microphones which are arranged such that the first microphone (12) receives a desired speech input and the background noise present in the vicinity of the speech, and the second microphone (14) receives substantially only the background noise. The background noise from the second microphone (14) is converted into a corresponding electrical signal and substracted (16) from a signal corresponding to the speech and background noise obtained from the first microphone (12) so as to produce a signal representing substantially the speech.

Two active element in-the-ear microphone system

Abstract: The herein described invention relates to an ear mounted, in-the-ear compound microphone system which simultaneously uses both an accelerometer, or vibration transducer, to sense bone conducted low speech frequencies and a microphone with controlled frequency response to sense airborne high speech frequencies within the ear canal. It combines the speech spectrum components picked up by the two transducers into a single composite audio signal with improved human speech frequency response characteristics. It simultaneously demonstrates significantly reduced sensitivity to surrounding background noise and provides measurable hearing protection for the user. Through adjustment and alteration of the supporting electronic circuitry, the operating characteristics and performance of the compound microphone can be changed. The in-the-ear microphone system can be used with a two-way speech system by installing a miniature earphone element within a common housing with the microphone.

Transmitter-receiver having ear-piece type acoustic transducing part

Abstract: Ear-piece type acoustic transducing part is provided with a bone-conducted sound pickup microphone for picking up a bone-conducted sound, a directional microphone for picking up an air-conducted sound and an electro-acoustic transducer for transducing a received speech signal to a received speech sound. A transmitting-receiving circuit connected to the acoustic transducing part includes: a low-pass filter which permits the passage therethrough of low-frequency components in a bone-conducted sound signal from the bone-conducted sound pickup microphone; a high-pass filter which permits the passage therethrough of high-frequency components in an air-conducted sound signal from the directional microphone; first and second variable loss circuits which impart losses to the outputs from the low-pass filter and the high-pass filter, respectively; a comparison/control circuit which compares the output levels of the low-pass filter and the high-pass filter with predetermined first and second reference levels, respectively, and based on the results of comparison, controls losses that are set in the first and second variable loss circuits; and a combining circuit which combines the outputs from the first and second variable loss circuits into a speech sending signal.

Headset for hands-free wireless telephone

Abstract: A wireless telephone using an RF receiver unit and an RF transmitter unit and having a removable headset, the telephone and headset particularly suitable for applications where the RF transmitter unit is switched on and off at a frequency less than twenty kilohertz. The wireless telephone is coupled to the headset using a conventional stereo plug and jack. A headset detect logic signal is generated from the speaker audio line. A microphone current detect logic signal is generated from the microphone audio line. A digital logic circuit, preferably including a microprocessor, monitors the headset detect logic signal and microphone current detect logic signals and generates a programmed response to the headset detect logic signal. The programmed response reduces audible noise to the headset user, generates a ringing signal on the speaker audio line when an incoming call is received and allows the headset user to answer incoming calls, and place outgoing calls while touching only the headset. The wireless telephone and headset include circuitry to reject RF noise and circuitry to attenuate RF noise passed into the wireless telephone by a long multi-conductor wire coupling the headset to the wireless telephone.

Method and apparatus for dynamic sound optimization

Abstract: This invention comprises a system to compensate for the noise level inside a vehicle by measuring the music level and the noise level in the vehicle through the use of analog to digital conversion and adaptive digital filtering, including a sensing microphone in the vehicle cabin to measure both music and noise; preamplification and analog to digital (A/D) conversion of the microphone signal; A/D conversion of a stereo music signal; a pair of filters that use an adaptive algorithm such as the known Least Mean Squares ("LMS") method to extract the noise from the total cabin sound; an estimation of the masking effect of the noise on the music; an adaptive correction of the music loudness and, optionally, equalization to overcome the masking effect; digital to analog (D/A) conversion of the corrected music signal; and transmission of the corrected music signal to the audio system.

A simple adaptive first-order differential microphone

Abstract: As communication devices become more portable and used in any environment, the acoustic pick-up by electroacoustic transducers will require the combination of small compact transducers and signal-processing to allow high quality communication. This paper covers the design and implementation of a novel adaptive first-order differential microphone. The self-optimization is based on minimizing the microphone output under the constraint that the solitary null for first order systems is located in the rear-half plane. The constraint is simply realized by the judicious subtraction of time-delayed outputs from two closely-spaced omnidirectional microphones. Although the solution presented does not maximize the signal-to-noise ratio, it can significantly improve the signal-to-noise ratio in certain acoustic fields.

Apparatus and methods for determining the relative displacement of an object

Abstract: A microphone is disclosed which converts an audio signal directly into a digital representation by analyzing and digitizing the distortion imposed upon a signal, such as a string of regularly spaced pulses as a result of the displacement of a diaphragm, relative to a sensor, in response to the incoming acoustical signal. Other devices, systems and methods are also disclosed.

Noise cancelling systems

Abstract: A first major aspect of the invention relates to an active noise cancelling system which detects ambient noise and applies electro-accoustic processing thereto to produce an acoustic signal for cancelling out the ambient noise. The active noise cancelling system may be used to cancel all noise but an audio signal which is desired to be heard by the user. A second major aspect of the invention relates particularly to a system for electronically cancelling noise input to a user microphone, by utilizing known characteristics of speech signal and ambient noise. The first major aspect of the invention can be used in concert with the second major aspect in a communication system at the earpiece of a telephone and the mouthpiece of a telephone.

A framework for speech source localization using sensor arrays

Abstract: Electronically steerable arrays of microphones have a variety of uses in speech data acquisition systems. Applications include teleconferencing, speech recognition and speaker identification, sound capture in adverse environments, and biomedical devices for the hearing impaired. An array of microphones has a number of advantages over a single-microphone system. It may be electronically aimed to provide a high-quality signal from a desired source location while simultaneously attenuating interfering talkers and ambient noise, does not necessitate local placement of transducers or encumber the talker with a hand-held or head-mounted microphone, and does not require physical movement to alter its direction of reception. Additionally, it has capabilities that a single microphone does not; namely automatic detection, localization, and tracking of active talkers in its receptive area. A fundamental requirement of sensor array systems is the ability to locate and track a speech source. An accurate fix on the primary talker, as well as knowledge of any interfering talkers or coherent noise sources, is necessary to effectively steer the array. Source location data may also be used for purposes other than beamforming; e.g. aiming a camera in a video-conferencing system. In addition to high accuracy, the location estimator must be capable of a high update rate as well as being computationally non-demanding in order to be useful for real-time tracking and beamforming applications. This thesis addresses the specific application of source localization algorithms for estimating the position of speech sources in a real room environment given limited computational resources. The theoretical foundations of a speech source localization system are presented. This includes the development of a source-sensor geometry for talkers and sensors in the near-field environment, the evaluation of several error criteria available to the problem, and the detailing of source detection and estimate-error prediction methods. Several practical algorithms necessary for real-time implementation are then developed, specifically the derivation and evaluation of an appropriate time-delay estimator and a novel closed-form locator. Finally, results obtained from several real systems are presented to illustrate the effectiveness of the proposed source localization techniques as well as to confirm the practicality of the theoretical models.

Magnetically controllable hearing aid

Abstract: An apparatus for controlling an adjustable operational parameter of a hearing aid by the use of an external magnetic actuator held in proximity with the hearing aid. The hearing aid has a microphone for generating signals, hearing aid circuitry for processing the signals, an output transducer for transforming the processed signals to a user compatible form, and a single magnetic switch, such as a reed switch, connected to the hearing aid circuitry. The magnetic switch controls the hearing aid circuitry to adjust an adjustable operational parameter, such as volume. In one embodiment the adjustable operational parameter continues to adjust or cycle between a minimum and a maximum as long as the magnetic actuator is maintained in proximity with the magnetic switch. When the magnetic actuator is removed the adjustment ceases. The invention allows precise adjustment and control of an adjustable parameter with minimal effort and movement by the user. The hearing aid circuitry may include a memory to allow a desired setting of the adjustable operational parameter to be saved when the hearing aid is turned off.

Sleep apnea screening and/or detecting apparatus and method

Abstract: Sleep apnea screening and/or detection apparatus for use by a patient breathing through the nose and/or mouth and producing an air flow into and out of the lungs of the patient and creating audible sounds. It is comprised of a first microphone positioned in the vicinity of the patient's nose and mouth and out of contact therewith to pick up audible sounds created by breathing of the patient and providing a first electrical analog signal. A second microphone is provided which is positioned near the patient for picking up ambient noise in the vicinity of the patient and providing a second electrical analog signal, an active noise controller is provided for combining the first and second electrical analog signals to provide a third electrical signal for generating a fourth electrical signal providing a waveform which is closely correlated with the air flow of the patient. A classifier is provided for classifying the electrical waveform provided by the fourth electrical signal to determine when a disordered breathing event has occurred.

Method and apparatus for multi-channel acoustic echo cancellation

Abstract: In a multi-channel acoustic echo cancellation, received signals in a plurality of channels are radiated as acoustic signals by a plurality of loudspeakers, received signal vectors in these channels are combined into a combined vector and a rearranged received signal vector in the case of at least two channels being exchanged is generated. By inputting the combined received signal vector into an echo replica generating part which simulates echo paths from the loudspeakers to at least one microphone, an echo replica is generated. The echo replica is subtracted from an echo output from the microphone to obtain a residual echo. Based on the relationship between the received signal vector and the corresponding residual echo and between the rearranged received signal vector and the corresponding approximated residual echo, an adjustment vector is obtained which is used to adjust the estimated echo path vector representing an impulse response of the echo replica generating part.

Speech circuit controlling sidetone signal by background noise level

Abstract: A speech circuit is disclosed which solves the serious problem of the degradation of the articulation of received speech voice in conventional circuits and permits pleasant communications at places where the background noise level is high. The circuit has a construction in which an input signal from a microphone is attenuated in correspondence to the background noise level to form a sidetone signal and a received speech signal from a speech channel is amplified in correspondence to the background noise level to form a new received speech signal.

Implantable microphone for cochlear implants and the like

Abstract: A system for assisting a person with a hearing disability includes an implantable microphone (40A) which senses sounds by monitoring pressure variations in the cochlear fluid (36). The electrical signal generated by the microphone (40A) is processed and used by a signal generator (44), such as a cochlear implant to generate excitation signals for the patient. In this manner, external microphones, used in prior art systems, are eliminated.

System and method for measuring acoustic reflectance

Abstract: A system and method of measuring the linear and nonlinear response of an unknown acoustic termination uses a small probe assembly containing a sound source and microphone to determine the reflection function of the unknown acoustic termination. The probe assembly is used with a calibration tube to calculate an electrical signal that will provide a desired acoustic stimulus signal to the acoustic termination. The calibration tube is also used to characterize the signal processing properties of the sound source and microphone, as well as other associated signal processing circuits such as amplifiers, filters, and the like. The calibrated system is subsequently coupled to the unknown acoustic termination to deliver the acoustic stimulus signal. The reflection function is indicative of the power transferred to the unknown acoustic termination. The measurement of the linear transfer characteristic is applicable to any unknown acoustic termination such as a musical instrument or the auditory system. The probe assembly is sized to be positioned directly within the outer portion of the ear and measure the linear characteristics of the ear. The system is further able to measure the nonlinear transfer characteristics of the ear by measuring the linear response at multiple levels of the acoustic stimulus. The system is particularly useful in testing the response of the middle ear and inner ear of humans or other animals.

Evaluation of a portable two-microphone adaptive beamforming speech processor with cochlear implant patients

Abstract: A two‐microphone noise reduction technique was tested with four cochlear implant patients. The noise reduction technique, known as adaptive beamforming (ABF), used signals from only two microphones—one behind each ear—to attenuate sounds not arriving from the direction directly in front of the patient. The algorithm was implemented in a portable digital signal processor, and was compared with a strategy in which the two microphone signals were simply added together (two‐microphone broadside strategy). Tests with the four patients were conducted in a soundproof booth with target speech arriving from in front of the patient and multitalker babble noise arriving at 90 deg to the left. Results at 0‐dB signal‐to‐noise level (S/N) showed large improvements in speech intelligibility for all patients, when compared to the two‐microphone broadside strategy. Precautions were taken to avoid cancellation of the target speech, and, accordingly, subjective tests showed no deterioration in performance for the adaptive beamformer in quiet. Physical measurement of the directional characteristics of the ABF was made with the microphones placed behind the ears of a KEMAR manikin and in the same acoustic environment as used with the patients. Results showed directional gain of approximately 10 dB when the angle of incidence for interfering noise was shifted more than 20 to 30 deg from directly in front of or behind the manikin. The effects of reverberation were explored by placing the manikin in different acoustical environments and observing the attenuation of the noise alone at various angles. A near anechoic environment allowed the noise to be attenuated by as much as 21 dB as the angle of incidence of the noise shifted away from directly in front of or behind the manikin. In a highly reverberant concrete stairwell, the adaptive beamformer was, however, unable to provide any directional gain beyond about 3 dB for any angle of incidence.

Two-way communications earset

Abstract: A one-piece two-way voice communication earset is situated in or at the ear of the user and is operated hands-free. It includes either two separated microphones having their outputs combined or a single bidirectional microphone. In either case, the earset treats the user's voice as consisting of out-of-phase signals that are not canceled, but treats ambient noise, and any incidental feedback of sound from received voice signals, as consisting of signals more nearly in-phase that are canceled or greatly reduced in level. The microphones are preferably electret microphones. In addition, a logarithmic amplification circuit reduces residual ambient noise during non-speaking intervals. The earset includes a receiver coupled to a communication medium and a speaker coupled to the receiver for emitting sound from received voice signals primarily to the user's ear canal. A transmitter, a part of the earset also coupled to the communication medium and located outside the ear canal, houses the two microphones or the one bidirectional microphone.

Method and apparatus including microphone arrays and neural networks for speech/speaker recognition systems

Abstract: A neural network is trained to transform distant-talking cepstrum coefficients, derived from a microphone array receiving speech from a speaker distant therefrom, into a form substantially similar to close-talking cepstrum coefficients that would be derived from a microphone close to the speaker, for providing robust hands-free speech and speaker recognition in adverse practical environments with existing speech and speaker recognition systems which have been trained on close-talking speech.

Low-feedback compact wireless telephone

Abstract: The present invention is directed to apparatus and methods for reducing sidetone feedback in a portable wireless telephone system by positioning the speaker and the microphone to reduce acoustic feedback and by controlling the amplifier gain in the transmitter section of the wireless telephone to reduce amplification of sidetone which has been acoustically coupled from the speaker to the microphone. The wireless telephone may be configured as a watch with a wrist strap for retaining it to the user, as a piece of jewelry such as a brooch or pendant, as a voice capable pager/intercom, or as an identification badge for an employee. In one embodiment, the base station includes circuitry for further reducing sidetone feedback. In additional embodiments, the wireless telephone may include a quick-release outboard battery, a retractable earpiece, and an ergonomic keypad.

Active noise reduction headset

Abstract: An active noise reduction headset including a headband and at least one earcup secured thereto. A microphone is mounted within the earcup for detecting and transducing acoustic pressure within the earcup to a corresponding microphone electronic signal. An electronic signal processing unit is mounted within the earcup and coupled with the microphone for generating an anti-noise signal from the microphone electronic signal. A first speaker is mounted within the earcup for receiving and acoustically reproducing an electronic anti-noise signal from the electronic signal processing unit. A second speaker is mounted within the earcup for receiving and acoustically reproducing an electronic communication signal.

Radio-based hearing aid system

Abstract: A personal radio-based hearing aid system is provided. The hearing aid system interfaces with existing hearing aids using the “T” facility. The system comprises a switchable unidirectional or omnidirectional microphone; a line input; and an FM radio transmitter. The components are housed in a discrete hand-held unit with integral stand, and a FM receiver unit connected to an inductive loop to form a discrete pendant and necklace. The prime use of said system is to give the user greater control over his environment by using the system to compensate for the loss of natural aural focus. The system addresses the problems of a “background noise” by capturing the desired sounds by selection

Active control of axial-flow fan noise

Abstract: An investigation was conducted using active noise control to reduce noise from small axial‐flow fan units commonly found in computers and printers. The fan unit itself was used as the cancellation source in the active noise control scheme, achieved by axially modulating the unit with a shaker. Feasibility studies which looked at radiation efficiency and transfer function data identified fan units which would adequately perform as efficient, undistorted sources of noise when driven by particular shakers. Once a suitable shaker and fan combination was discovered, simulations of active noise control were conducted in MATLAB which utilized the measured error path impulse response (representing the system which defines the output voltage response of a microphone near the fan to an input voltage supplied to the shaker). Results from the simulation showed that an experiment could be constructed which would effectively reduce the tonal components from the fan unit. An experimental demonstration was constructed, results from which show a 20‐dB reduction in sound‐pressure level for the blade passage tone, a 15‐dB reduction for the second harmonic, and a 7‐ to 8‐dB reduction for the third harmonic. [Work supported by IBM through Shared University Research Program.]

Knowing who to listen to in speech recognition: visually guided beamforming

Abstract: With speech recognition systems steadily improving in performance, freedom from head-sets and push-buttons to activate the recognizer is one of the most important issues to achieve user acceptance. Microphone arrays and beamforming can deliver signals that suppress undesired jamming signals but rely on knowledge where the signal is in space. This knowledge is usually derived by identifying the loudest signal source. Knowing who is speaking to whom and where should however not depend on loudness, but on the communication purpose. In this paper, we present acoustic and visual modules that use tracking of the face of a speaker of interest for sound source localization and beamforming for signal extraction. It is shown that in noisy environments a more accurate localization in space can be delivered visually than acoustically. Given a reliable location finder, beamforming substantially improves recognition accuracy.

Coupled adaptive filters for acoustic echo control and noise reduction

Abstract: Presents new adaptive algorithms for acoustic echo control and noise reduction which employ one, two, or possibly more microphone signals. The new algorithms accommodate high echo attenuation and lead to implementations with reduced complexity. These algorithms combine a conventional FIR echo canceller with a second NLMS-adapted FIR filter which attenuates residual echoes. The paper presents a one-microphone system with improved echo attenuation and a two-microphone system which attenuates acoustic echoes as well as ambient noise and near end speech reverberation. The algorithms can be interpreted as a frequency selective generalization of the well known voice controlled switch. The paper explains the algorithms and presents experimental results in real acoustic environments.

Modeling, identification and feedback control of noise in an acoustic duct

Abstract: We consider a single-input, single-output plant involving one control actuator (speaker) and one control sensor (microphone). Additional speakers and microphones are used to provide disturbances and to assess closed-loop performance. To simplify matters, we confine our consideration in this paper to the case of a collocated sensor and actuator, that is, the control speaker and control microphone located at the same position along the duct. This configuration has been studied in the noise control literature under the name of tightly coupled monopole. In designing feedback controllers for the acoustic duct, we apply modern state space control techniques. The use of such techniques is necessitated by the high order of the identified model, which, for a 400 Hz modeling bandwidth in our experiment, involves 30 states. Feedback controllers designed for noise suppression were obtained by applying LQG synthesis with suitable precompensation to assure robustness to high frequency uncertainty.

Noise-canceling differential microphone assembly

Abstract: Improved microphone performance is achieved by configuring a second-order derivative microphone assembly in such a way that radially divergent near-field input produces a microphone response proportional to a first-order spatial derivative of the acoustic pressure field.

Matched-filter processing of microphone array for spatial volume selectivity

Abstract: Performance of a delay-and-sum beamformer is typically degraded in a reverberant enclosure because the beam captures not only the desired source signal (direct path) but also all images along the beam axis. Matched-filter processing of microphone arrays is shown to improve the quality of sound capture in reverberant environments. This paper also reports the performance of matched-filter array processing in a real room. Significant improvements are shown over single microphones and the traditional delay-and-sum beamformer.