Showing papers on "Microphone published in 2002"

Noise suppression for speech signal in an automobile

Abstract: Techniques for suppressing noise from a signal comprised of speech plus noise. A first signal detector (e.g., a microphone) provides a first signal comprised of a desired component plus an undesired component. A second signal detector (e.g., a sensor) provides a second signal comprised mostly of an undesired component. The adaptive canceller removes a portion of the undesired component in the first signal that is correlated with the undesired component in the second signal and provides an intermediate signal. The voice activity detector provides a control signal indicative of non-active time periods whereby the desired component is detected to be absent from the intermediate signal. The noise suppression unit suppresses the undesired component in the intermediate signal based on a spectrum modification technique and provides an output signal having a substantial portion of the desired component and with a large portion of the undesired component removed.

Microelectromechanical system package with environmental and interference shield

Abstract: A microelectromechanical system package has a microelectromechanical system microphone, a substrate, and a cover. The substrate has a surface for supporting the microelectromechanical microphone. The cover includes a conductive layer having a center portion bounded by a peripheral edge portion. A housing is formed by connecting the peripheral edge portion of the cover to the substrate. The center portion of the cover is spaced from the surface of the substrate to accommodate the microelectromechanical system microphone. The housing includes an acoustic port for allowing an acoustic signal to reach the microelectromechanical system microphone.

Sound enhancement for mobile phones and other products producing personalized audio for users

Abstract: A mobile phone or other personal communication device includes resources applying measures of an individual's hearing profile, personal choice profile, and induced hearing loss profile, separately or in combination, to build the basis of sound enhancement. A personal communication device thus comprises a transmitter/receiver coupled to a communication medium for transmitted receiving audio signals, control circuitry that controls transmission, reception and processing of call and audio signals, a speaker, and a microphone. The control circuitry includes logic applying one or more of a hearing profile of the user, a user preference related hearing, and environmental noise factors in processing the audio signals. The control circuitry may includes instruction memory and an instruction execution processor such as a digital signal processor.

Hearing aid and a method for enhancing speech intelligibility

Abstract: A hearing aid (22) having a microphone (1), a processor (53) and an output transducer (12), is adapted for obtaining an estimate of a sound environment, determining an estimate of the speech intelligibility according to the sound environment estimate, and for adapting the transfer function of the hearing aid processor in order to enhance the speech intelligibility estimate. The method according to the invention achieves an adaptation of the processor transfer function suitable for optimizing the speech intelligibility in a particular sound environment. Means for obtaining the sound environment estimate and for determining the speech intelligibility estimate may be incorporated in the hearing aid processor, or they may be wholly or partially implemented in an external processing means (56), adapted for communicating data to the hearing aid processor via an appropriate link.

Beamforming In Acoustic Testing

Abstract: Non-intrusive measurements of wind tunnel model noise can be made in non-acoustic, hardwall wind tunnels, as well as free jets and other challenging settings, by using beamforming techniques with sparse, wide-band phased arrays of microphones The process, which was developed for this application at Boeing in the last few years, begins by measuring a full array cross-spectral matrix for each frequency of interest A grid of potential source locations in the test section is defined A complex array steering vector is computed for each grid point with non-uniform flow, microphone imperfections, and installation effects taken into account Beamforming combines the cross-spectral matrices and the steering vectors to produce maps of the wind tunnel model’s acoustic source distribution Several beamforming algorithms are described with intuitive motivation and expected performance Practical details of steering vector construction are presented Techniques for removing various types of interference are detailed

Real time display of system instructions

Abstract: A system and method for reviewing inputted voice instructions in a vehicle-based telematics control unit. The system includes a microphone, a speech recognition processor, and an output device. The microphone receives voice instructions from a user. Coupled to the microphone is the speech recognition processor that generates a voice signal by performing speech recognition processing of the received voice instructions. The output device outputs the generated voice signal to the user. The system also includes a user interface for allowing the user to approve the outputted voice signal, and a communication component for wirelessly sending the generated voice signal to a server over a wireless network upon approval by the user.

A beamforming approach to permutation alignment for multichannel frequency-domain blind speech separation

Abstract: In this work, we explore important connections between blind source separation (BSS) and ideal beamforming. We first compare the performance of a null-steering beamformer against that of a frequency-domain BSS method in a reverberant environment, drawing some interesting conclusions. We then examine the feasibility of using beamformer concepts to resolve permutation inconsistency across frequency, which degrades the performance of BSS methods in a reverberant environment. We also propose a permutation alignment scheme based on information gathered from the microphone array directivity patterns. This technique is novel in the sense that it works satisfactorily even when the directivity patterns exhibit grating lobes, where, in fact, better separation can be achieved in principle. We perform experiments that support the viability of the proposed method under different operating conditions and microphone spacings.

Silicon-based transducer for use in hearing instruments and listening devices

Abstract: A silicon-based transducer assembly coupled to a movable structure in a hearing instrument. The transducer assembly includes at least one microphone chip and an ASIC having multiple integrated components such as any combination of a DSP, an A/D converter, an amplifier, a filter, or a wireless interface. The movable structure may be a battery access door, a volume dial, a switch, or a touch pad. A protection strip can be disposed across the battery access door to prevent debris from clogging the silicon-based transducer assembly. The transducer assembly may also include an array of microphone chips to achieve adaptive beam steering or directionality. When equipped with a wireless interface, the hearing instrument wirelessly communicates with another hearing instrument or with a network.

Method and apparatus for communication operator privacy

Abstract: Disclosed is a device for use with communications equipment to provide privacy for the operator and reduce acoustic noise from the operator's voice in the area. The device uses active acoustic cancellation to silence the voice of the operator once past, and captured by the microphone. Embodiments include the all types of microphones for any type of telephone, transmitting radio, intercom or other communication devices where a operator speaks out loud to communicate with another location.

Vehicle accessory microphone

Abstract: A microphone assembly (1700) includes one or more transducers (1702) positioned in a housing (1732). Circuitry (Fig. 8) is coupled to the transducer for outputting an electrical signal such that the microphone has a main lobe directed forwardly and attenuates signals originating from the sides and/or rear. The transducers can advantageously include multiple transducers, which, with the circuit, produce a desired sensitivity pattern. The microphone assembly can be employed in a vehicle accessory.

An integrated audio and active noise control headsets

Abstract: This paper presents an integrated approach in designing a noise reduction headset for the audio and communication applications. Conventional passive headsets give good attenuation of ambient noise in the upper frequency range, while most of these devices fail below 500 Hz. Unlike the feedforward method, the adaptive feedback active noise control technique provides a more accurate noise cancellation since the microphone is placed inside the ear-cup of the headset. Furthermore, the system uses single microphone per ear cup, thus produces a more compact, lower power consumption, cheaper solution, and ease of integration with existing audio and communication devices to form an integrated feedback active noise control headsets. Simulation results have been conducted to show that the integrated approach can remove the disturbing noise and at the same time, allow the desired speech or audio signal to pass through without cancellation.

Microphone elements for a computing system

Abstract: An improved speech recognition device is provided. The speech recognition device comprises a display (204) with at least two built in microphones (208, 212) and a speech recognition module (202) electrically connected to the display. The speech recognition module uses an algorithm that may take into account the position of the built in microphone on the display. The display may have a first axis of rotation where the microphones may be placed an equal distance from the first axis of rotation.

Improved microphone array configurations for auralization of sound fields by Wave Field Synthesis.

Abstract: In order to correctly reproduce ( “auralize” ) the acoustic wave field in a hall through a wave-field synthesis (WFS) system, impulse responses are nowadays measured along arrays of microphone positions. Three array configurations are considered ‐ linear, cross, and circular. The linear and cross array configurations both have strong limitations, most of which can be avoided by using circular arrays. Auralization techniques are explained for all types of arrays. For the circular array configuration the connection between circular holophony, highorder incoming and outgoing ambisonics, and plane-wave decomposition for a sound field will be established and used as a tool for auralization.

System and method for speech enhancement

Abstract: A method and apparatus for reducing noise in a speech signal. A handset or remote unit provides to users with a hearing deficiency, a first mode of operation where noise suppressant/speech enhancement algorithms are used during any auditory-related service. There is also provided, in a related mode of operation, speech filtering for reducing noise in a speech signal received through the microphone and outputting the filtered sound to the speaker. The handset includes a microphone for receiving an auditory sound, a receiver for receiving an auditory signal and a speech filter for suppressing noise in the auditory signal and sound. The speech filter also may be configured to shift the frequency and/or alter the intensity of the auditory signal and sound. The speaker is used for amplifying and outputting the enhanced speech component as an audible sound.

Digital hearing aid system

Abstract: A digital hearing aid is provided that includes front and rear microphones, a sound processor, and a speaker. Embodiments of the digital hearing aid include an occlusion subsystem, and a directional processor and headroom expander. The front microphone receives a front microphone acoustical signal and generates a front microphone analog signal. The rear microphone receives a rear microphone acoustical signal and generates a rear microphone analog signal. The front and rear microphone analog signals are converted into the digital domain, and at least the front microphone signal is coupled to the sound processor. The sound processor selectively modifies the signal characteristics and generates a processed signal. The processed signal is coupled to the speaker which converts the signal to an acoustical hearing aid output signal that is directed into the ear canal of the digital hearing aid user. The occlusion sub-system compensates for the amplification of the digital hearing aid user's own voice within the ear canal. The directional processor and headroom expander optimizes the gain applied to the acoustical signals received by the digital hearing aid and combine the amplified signals into a directionally-sensitive response.

Directional sound acquisition

Abstract: Directional sound acquisition is obtained by combining directional sensitivities in microphones with signal processing electronics to reduce the effects of noise received from unwanted directions. One or more microphones having directional sensitivity including a minor lobe pointing in the particular direction of interest and a major lobe pointing in a direction other than the particular direction are used. Signal processing circuitry reduces the effect of sound received from directions of a microphone major lobe.

Bone conduction hearing aid

Abstract: A bone conduction hearing aid (10) includes a vibration (16) carried by the insertion end (14) of the hearing aid (10). When the hearing aid (10) is inserted into the ear canal (12) of a patient, the vibrator (16) is positioned in the ear canal (12) adjacent the mastoid bone (18). A microphone (24) receives sound waves and outputs a microphone signal to the hearing aid electronics (34) where the microphone signal is amplified and then sent to the vibrator (16), causing the vibrator (16) to vibrate. Vibrations produced by the vibrator (16) are transferred to the opposite cochlea by way of the mastoid bone (18), enabling enhanced hearing perception in patients with hearing loss in one ear. Transfer of vibrations to the bones of the middle ear also assists patients with conductive pathology in one ear. The hearing aid (10) may also function to enhance communication in high noise environments. Feedback from the vibrator (16) to the microphone (24) is eliminated electronically. Various alternate forms of feedback elimination are also contemplated by the invention.

Performance of directional microphone hearing aids in everyday life.

Abstract: This study explored the use patterns and benefits of directional microphone technology in real-world situations experienced by patients who had been fitted with switchable omnidirectional/directional hearing aids. Telephone interviews and paper-and-pencil questionnaires were used to assess perceived performance with each microphone type in a variety of listening situations. Patients who used their hearing aids regularly and switched between the two microphone configurations reported using the directional mode, on average, about one-quarter of the time. From brief descriptions, patients could identify listening situations in which each microphone mode should provide superior performance. Further, they reported encountering listening situations in which an omnidirectional microphone should provide better performance more frequently than listening situations in which the directional microphones should be superior. Despite using the omnidirectional mode more often and encountering situations in which an omnidirectional microphone should provide superior performance more frequently, participants reported the same level of satisfaction with each microphone type.

Self-localizing dynamic microphone arrays

Abstract: This paper introduces a mechanism for localizing a microphone array when the location of sound sources in the environment is known. Using the proposed spatial observability function based microphone array integration technique, a maximum likelihood estimator for the correct position and orientation of the array is derived. This is used to localize and track a microphone array with a known and fixed geometrical structure, which can be viewed as the inverse sound localization problem. Simulations using a two-element dynamic microphone array illustrate the ability of the proposed technique to correctly localize and estimate the orientation of the array even in a very reverberant environment. Using 1 s male speech segments from three speakers in a 7 m by 6 m by 2.5 m simulated environment, a 30 cm inter-microphone distance, and PHAT histogram SLF generation, the average localization error was approximately 3 cm with an average orientation error of 19/spl deg/. The same simulation configuration but with 4 s speech segments results in an average localization error less than 1cm, with an average orientation error of approximately 2/spl deg/. Experimental examples illustrate localizations for both stationary and dynamic microphone pairs.

Microphone array signal enhancement

Abstract: A system and method facilitating signal enhancement utilizing an adaptive filter is provided. The invention includes an adaptive filter that filters an input based upon a plurality of adaptive coefficients and modifies the adaptive coefficients based on a feedback output. A feedback component provides the feedback output based, at least in part, upon a non-linear function of the acoustic reverberation reduced output. Optionally, the system can further include a linear prediction (LP) analyzer and/or a LP synthesis filter. The system can enhance signal(s), for example, to improve the quality of speech that is acquired by a microphone by reducing reverberation. The system utilizes, at least in part, the principle that certain characteristics of reverberated speech are measurably different from corresponding characteristics of clean speech. The system can employ a filter technology (e.g., reverberation reducing) based on a non-linear function, for example, the kurtosis metric.

Interactive toy system

Abstract: An interactive toy has a microphone, a speaker, a memory for storing a toy identifier, and an interface to provide communications with a computer system. The computer system connects to a server on a network. The interactive toy provides electrical signals from the microphone, as well as the toy identifier, to the computer system via the interface. The interface enables the computer system to control the speaker to generate audible information according to data received from the server. Alternatively, a processor and memory with networking capabilities may be embedded within the toy to eliminate the need for a computer system.

Spacing for microphone elements

Abstract: A speech recognition device with a frequency range with an upper frequency limit fmax is provided. The speech recognition device has more than two microphones with distances between the microphones, wherein the greatest common factor of the distances between the microphones is less than the speed of sound divided by fmax. More particularly, where the microphones are spaced a total distance, the number of the more than two microphones is less than the one half the total distance times the upper frequency limit divided by the speed of sound.

Microphone with improved sound inlet port

Abstract: A microphone comprises a housing defining an inner volume and including a first exterior surface with an aperture leading to the inner volume The microphone includes a transducing assembly within the housing for converting sound into an electrical signal A sound inlet plate defines, typically in combination with the first exterior surface, a passageway for transmitting sound to the aperture The passageway receives the sound from an opening in the sound inlet plate The opening is offset from the location at which the aperture is positioned on the exterior surface The sound inlet plate is made very thin so that it does not extend substantially away from the housing

Cellular phone with high-quality sound reproduction capability

Abstract: In a cellular phone, stereophonic sound reproduction of music etc. is realized by using a microphone and a speaker of the cellular phone as stereophonic speakers. By the stereophonic sound reproduction function, radio broadcasting from FM stations, streaming sound from the Internet, etc. are reproduced by the cellular phone. The speaker and the microphone are removed from the front of the housing of the cellular phone (where a display and operation buttons are mounted) and are placed on the back of the housing, thereby the diameters of the speaker and the microphone are enlarged and thereby wide-range and high-volume-level sound reproduction is realized. The speaker and the microphone are preferably implemented by low-profile flat-panel speakers that are functionally equivalent. Further, by the removal of the speaker and microphone from the front of the housing, the sizes of the display and the operation buttons are enlarged, thereby the viewability of the display and the operability of the operation buttons are improved. Such features are effectively realized in, for example, a folding cellular phone that can be unfolded to an angle larger than 180°.

A spherical microphone array for spatial sound recording

Abstract: The progression of audio from monophonic to the present day 5‐channel playback is being driven by the desire to improve the immersion of the listener into the acoustic scene. In the limit, the goal is the reconstruction of the original sound field. This talk describes the decomposition of the sound field into orthogonal components, the so‐called spherical harmonics and is directly related to the method of Ambisonics. These components contain all required information to allow a reconstruction of the original sound field. This approach is scalable to any number of loudspeakers and is also backwards compatible to surround sound, stereo and mono playback. One problem is the recording of the orthogonal components. So far only solutions exist that allow the recording of spherical harmonics up to first order. This limits the spatial resolution. This presentation introduces a new microphone that overcomes this limitation. It consists of pressure sensors that are equally distributed on the surface of a rigid sphere. The number of sensors depends on the highest order spherical harmonic to be recorded. A minimum of (n+1)2 sensors is required to record harmonics up to nth order. The sensor signals are then processed to give the desired spherical harmonic outputs.

Method of speech recognition resistant to convolutive distortion and additive distortion

Abstract: A speech recognizer operating in both ambient noise (additive distortion) and microphone changes (convolutive distortion) is provided. For each utterance to be recognized the recognizer system adapts HMM mean vectors with noise estimates calculated from pre-utterance pause and a channel estimate calculated using an Estimation Maximization algorithm from previous utterances.

Influence of environmental factors on hearing aid microphone preference.

Abstract: The purpose of this study was to identify characteristics of everyday listening situations that influence user preferences for omnidirectional versus directional hearing aid microphones . Eleven experienced hearing aid users were fitted with digital hearing aids featuring switchable omnidirectional (OMNI) and adaptive-directional (DIR) modes (programs). For 6 weeks, their task was to identify and describe at least one listening situation each day in which one program performed better than the other using a checklist daily journal format. All participants reported difficulty identifying situations in which they could perceive a difference between the two microphone modes . Although an equal number had been requested, descriptions favoring the DIR outnumbered those for the OMNI . Chi-square tests were used to compare the distributions of 60 descriptions favoring the OMNI and 155 favoring the DIR across variables associated with the primary talker to whom the hearing aid user was listening, background noise, and other environmental characteristics . The results indicated that location of the primary talker, presence or absence and type of background noise, and type of space in which the communication occurred influenced microphone choice.

Voice acquisition system for a vehicle

Abstract: A voice acquisition system for a vehicle includes an interior rearview mirror assembly attached at an inner portion of the vehicle. The mirror assembly preferably includes at least one microphone for receiving audio signals within a cabin of the vehicle and generating an output signal indicative of these audio signals, the at least one microphone providing sound capture for at least one of a hands free cell phone system, an audio recording system and an emergency communication system. The system includes a control that receives the output signal from the at least one microphone and distinguishes the presence of vocal signals from non-vocal signals. The system preferably comprises a learning mode whereby a vocal characteristic of a particular driver or occupant of the vehicle is learnt so that the ratio of vocal signals emanating from that particular driver or occupant to non-vocal noise signals received by the system is enhanced.

Recording a three dimensional auditory scene and reproducing it for the individual listener

Abstract: A system for recording and reproducing a three dimensional auditory scene for individual listeners includes one or more microphone arrays (2 and 16); a support (3) for holding, moving the microphone array and also for attaching other devices (14); a data storage and encoding device (9); a control interface (13), and a processor and decoding device (10). The microphones in the microphone array (2) preferably have strong directional characteristics. The microphone array support mount (4) can support one or more physical structures (5) to provide directional acoustic filtering. The directional microphone array is electrically connected via a lead (8) to the sound encoding processor (9) and sound decoding processor (10). As the directional microphone array has acoustically directional properties, these properties can be adjusted using signal processing methods to match the acoustics of the external ears of the individual listener and thus result in a perceptually accurate recording and reproduction of a three dimensional auditory scene for the individual listener.

Cylindrical microphone having an electret assembly in the end cover

Abstract: A microphone includes a separate end cover with a sound port. A diaphragm is directly attached to the end cover. The backplate is positioned within the housing against a ridge near an end of the housing. A spacer is positioned against the backplate. The diaphragm engages the spacer when the end cover, with its attached diaphragm, is installed in the housing. The backplate of the microphone has an integral connecting wire that is made of the same material as the backplate. The integral connecting wire may have an inherent spring force to provide a pressure contact with the accompanying electrical components. The integral connecting wire electrically couples the backplate to the electronic components within the housing and transmits the raw audio signal corresponding to movement of the diaphragm. The housing may have first and second ridges on which the printed circuit board and the electret assembly are mounted, respectively.