Showing papers on "Microphone published in 2006"

Instability waves in a subsonic round jet detected using a near-field phased microphone array

Abstract: We propose a diagnostic technique to detect instability waves in a subsonic round jet using a phased microphone array. The detection algorithm is analogous to the beam-forming technique, which is typically used with a far-field microphone array to localize noise sources. By replacing the reference solutions used in the conventional beam-forming with eigenfunctions from linear stability analysis, the amplitudes of instability waves in the axisymmetric and first two azimuthal modes are inferred. Experimental measurements with particle image velocimetry and a database from direct numerical simulation are incorporated to design a conical array that is placed just outside the mixing layer near the nozzle exit. The proposed diagnostic technique is tested in experiments by checking for consistency of the radial decay, streamwise evolution and phase correlation of hydrodynamic pressure. The results demonstrate that in a statistical sense, the pressure field is consistent with instability waves evolving in the turbulent mean flow from the nozzle exit to the end of the potential core, particularly near the most amplified frequency of each azimuthal mode. We apply this technique to study the effects of jet Mach number and temperature ratio on the azimuthal mode balance and evolution of instability waves. We also compare the results from the beam-forming algorithm with the proper orthogonal decomposition and discuss some implications for jet noise.

Portable GPS navigation device

Abstract: A portable GPS navigation device displays map information and navigation instructions; the device (a) includes a two-way wireless transceiver operable to connect to a mobile telephone; (b) is operable to enable a user to control one or more functions of the mobile telephone and (c) includes an audio output and a microphone to enable a voice call to be made using the mobile telephone.

A two-stage algorithm for one-microphone reverberant speech enhancement

Abstract: Under noise-free conditions, the quality of reverberant speech is dependent on two distinct perceptual components: coloration and long-term reverberation. They correspond to two physical variables: signal-to-reverberant energy ratio (SRR) and reverberation time, respectively. Inspired by this observation, we propose a two-stage reverberant speech enhancement algorithm using one microphone. In the first stage, an inverse filter is estimated to reduce coloration effects or increase SRR. The second stage employs spectral subtraction to minimize the influence of long-term reverberation. The proposed algorithm significantly improves the quality of reverberant speech. A comparison with a recent enhancement algorithm is made on a corpus of speech utterances in a number of reverberant conditions, and the results show that our algorithm performs substantially better.

Active noise control system for headphone applications

Abstract: This paper presents the design and implementation of an adaptive feedback active noise control (ANC) system for headphone applications. The ideal position of the error microphone in the ear-cup was studied and determined experimentally, and music signals were used for adaptive system identification of the secondary path. The designed ANC headphone was implemented using the TMS320C32 digital signal processor for real-time experiments. Performance has been evaluated and compared with a high-end commercial ANC headphone using the same set of primary noises including real-world engine noises. Experiment results show the proposed ANC headphone achieves higher noise cancellation, especially for low-frequency harmonics

The Plenacoustic Function and Its Sampling

Abstract: The spatialization of the sound field in a room is studied, in particular the evolution of room impulse responses as a function of their spatial positions. It was observed that the multidimensional spectrum of the solution of the wave equation has an almost bandlimited character. Therefore, sampling and interpolation can easily be applied using signals on an array. The decay of the spectrum is studied on both temporal and spatial frequency axes. The influence of the decay on the performance of the interpolation is analyzed. Based on the support of the spectrum, the number and the spacing between the microphones is determined for the reconstruction of the sound pressure field up to a certain temporal frequency and with a certain reconstruction quality. The optimal sampling pattern for the microphone positions is given for the linear, planar and three-dimensional case. Existing techniques usually make use of room models to recreate the sound field present at some point in the space. The presented technique simply starts from the measurements of the sound pressure field in a finite number of positions and with this information the sound pressure field can be recreated at any spatial position. Finally, simulations and experimental results are presented and compared with the theory

Eyeglasses with wireless communication features

Abstract: A microphone, transmitter, speaker, receiver, and power source, all mounted to an eyeglasses frame, for sending and receiving signals wirelessly to and from a remote cell phone or other electronic device. The microphone and the transmitter can be mounted to extension arms that can be extended, pivoted, or otherwise moved to a position for use, and then moved to a stored position when not in use. Alternatively, the microphone, transmitter, speaker, receiver, and power source, can be mounted onto a clip-on or other attachment member that mounts onto a conventional eyeglasses frame, or to a hat or other article worn on the head.

Adaptation of a speech processing system from external input that is not directly related to sounds in an operational acoustic environment

Abstract: A speech processing system that performs adaptations based upon non-sound external input, such as weather input. In the system, an acoustic environment can include a microphone and speaker. The microphone/speaker can receive/produce speech input/output to/from a speech processing system. An external input processor can receive non-sound input relating to the acoustic environment and to match the received input to a related profile. A setting adjustor can automatically adjust settings of the speech processing system based upon a profile based upon input processed by the external input processor. For example, the settings can include customized noise filtering algorithms, recognition confidence thresholds, output energy levels, and/or transducer gain settings.

Automatic audio transducer adjustments based upon orientation of a mobile communication device

Abstract: A solution for automatically activating different audio transducers of a mobile communication device based upon an orientation of the device. In the solution, a series of speaker/microphone assemblies can be positioned on the device, such as positioned near an earpiece and positioned near a mouthpiece. Different speaker/microphone assemblies can also be positioned on the front of the device and on the back of the device. The solution can automatically determine an orientation for the device, based upon a detected direction of a speech emitting source and/or based upon one or more sensors, such as a tilt sensor and an accelerometer. For example, when a device is in an upside down orientation, an earpiece microphone and a mouthpiece speaker can be activated. In another example, an otherwise deactivated rear facing speaker can be activated when the device is oriented in a rear facing orientation.

Music headphones for manual control of ambient sound

Abstract: A headphone and method for allowing manually controlled reception of ambient sound by a user. More particularly, the invention relates to a headphone for listening to an audio signal while allowing varying amounts of ambient sound to the user. In one aspect, the headphone comprises a manually-controllable physical characteristic that controls the amount of ambient sound accessible to the user. The physical characteristic can be sound ports that can be opened or closed. In another aspect, the headphone can comprise a microphone receptive of ambient sound, wherein the user can control the proportion of sound that comes from the ambient sound or the audio signal. The microphone can be part of a noise cancellation system.

Computer communications using acoustic signals

Abstract: A method of communicating with an electronic device. The method includes providing an electronic device having an audible sound receiving and generating sub-system including a microphone, transmitting from a source at least one acoustic signal encoded with information, receiving said at least one acoustic signal by said microphone and determining a spatial position, distance or movement of the microphone relative to the source, responsive to the received at least one signal.

Micromachined microphone and multisensor and method for producing same

Abstract: A micromachined microphone is formed from a silicon or silicon-on-insulator (SOI) wafer. A fixed sensing electrode for the microphone is formed from a top silicon layer of the wafer. Various polysilicon microphone structures are formed above a front side of the top silicon layer by depositing at least one oxide layer, forming the structures, and then removing a portion of the oxide underlying the structures from a back side of the top silicon layer through trenches formed through the top silicon layer. The trenches allow sound waves to reach the diaphragm from the back side of the top silicon layer. In an SOI wafer, a cavity is formed through a bottom silicon layer and an intermediate oxide layer to expose the trenches for both removing the oxide and allowing the sound waves to reach the diaphragm. An inertial sensor may be formed on the same wafer, with various inertial sensor structures formed at substantially the same time and using substantially the same processes as corresponding microphone structures.

Acoustic source detection and localization based on wavefield decomposition using circular microphone arrays

Abstract: This paper is concerned with the problem of detecting and localizing multiple wideband acoustic sources by applying the notion of wavefield decomposition using circular microphone arrays optionally mounted into cylindrical baffles. The decomposed wavefield representation is used to serve as a basis for so-called modal array signal processing algorithms, which have the significant advantage over classical array signal processing algorithms that they inherently support multiple wideband acoustic sources. A rigorous derivation of modal array signal processing algorithms for source detection and localization, as well as performance evaluations, by means of measurements using an actual real-time capable implementation are presented.

Dual-Microphone Spatial Noise Suppression

Abstract: Spatial noise suppression for audio signals involves generating a ratio of powers of difference and sum signals of audio signals from two microphones and then performing noise suppression processing, e.g., on the sum signal where the suppression is limited based on the power ratio. In certain embodiments, at least one of the signal powers is filtered (e.g., the sum signal power is equalized) prior to generating the power ratio. In a subband implementation, sum and difference signal powers and corresponding the power ratio are generated for different audio signal subbands, and the noise suppression processing is performed independently for each different subband based on the corresponding subband power ratio, where the amount of suppression is derived independently for each subband from the corresponding subband power ratio. In an adaptive filtering implementation, at least one of the audio signals can be adaptively filtered to allow for array self-calibration and modal-angle variability.

System and method for utilizing inter-microphone level differences for speech enhancement

Abstract: Systems and methods for utilizing inter-microphone level differences to attenuate noise and enhance speech are provided. In exemplary embodiments, energy estimates of acoustic signals received by a primary microphone and a secondary microphone are determined in order to determine an inter-microphone level difference (ILD). This ILD in combination with a noise estimate based only on a primary microphone acoustic signal allow a filter estimate to be derived. In some embodiments, the derived filter estimate may be smoothed. The filter estimate is then applied to the acoustic signal from the primary microphone to generate a speech estimate.

Synchronized and noise-robust audio recordings during realtime magnetic resonance imaging scans.

Abstract: This letter describes a data acquisition setup for recording, and processing, running speech from a person in a magnetic resonance imaging (MRI) scanner. The main focus is on ensuring synchronicity between image and audio acquisition, and in obtaining good signal to noise ratio to facilitate further speech analysis and modeling. A field-programmable gate array based hardware design for synchronizing the scanner image acquisition to other external data such as audio is described. The audio setup itself features two fiber optical microphones and a noise-canceling filter. Two noise cancellation methods are described including a novel approach using a pulse sequence specific model of the gradient noise of the MRI scanner. The setup is useful for scientific speech production studies. Sample results of speech and singing data acquired and processed using the proposed method are given.

Hearing aid with large diaphragm microphone element including a printed circuit board

Abstract: A disposable-type hearing aid uses a relatively large single diaphragm or a large single diaphragm subdivided into a plurality of smaller active diaphragm areas obtained using a grate-like back support plate with ridges which contact and divide the diaphragm into the several smaller active diaphragm areas. The diaphragm and a backplate are enclosed in a metal housing and are disposed proximal and parallel to a shell-like hearing aid enclosure having sound inlets. The metal housing is closed at an end opposite the sound inlets by a printed circuit board (PCB) forming an acoustical seal for a back volume of the microphone. The PCB also carries substantially all the electronic components for the hearing aid thereon. The PCB has a ground plane in contact with the housing whereby the PCB also acts as an EMI shield. An electrical connection is formed in various ways between the back support plate and the PCB during assembly of the metal housing and components with the PCB. Mass production of disposable hearing aids with large diaphragms and relatively low noise levels is thus possible using this invention.

Detection and suppression of wind noise in microphone signals

Abstract: To reliably and consistently detect desirable sounds, a system detects the presence of wind noise based on the power levels of audio signals. A first transducer detects sound originating from a first direction and a second transducer detects sound originating from a second direction. The power levels of the sound are compared. When the power level of the sound received from the second transducer is less than the power level of the sound received from the first transducer by a predetermined value, wind noise may be present. A signal processor may generate an output from one or a combination of the audio signals, based on a wind noise detection.

System and method for generating auditory spatial cues

Abstract: This invention relates to a hearing aid system (100, 200, 300) for generating auditory spatial cues. The hearing aid system (100, 200, 300) comprises a first microphone unit (306) adapted to convert sound received at a first microphone (102) and received at a second microphone (104), a first delay unit (106) connected to the first microphone (102) delaying the signal from the first microphone (102), a first calculation unit (108) for summing the delayed signal of the first microphone (102) and signal of the second microphone (104), a processor unit (110) processing the summed signal, and a speaker converting the processed signal to a processed sound. The first and second microphones (102, 104) are separated by a predetermined first distance and the first delay unit (106) provides a predetermined first delay thereby generating a first auditory spatial cue representing a first spatial dimension in the summed signal.

Digital stethoscope and monitoring instrument

Abstract: An electronic stethoscope includes a microphone; an accelerometer to detect stethoscope movement; a processor coupled to the microphone and the accelerometer; and a speaker coupled to the processor to reproduce a biological sound.

The effect of multi-channel wide dynamic range compression, noise reduction, and the directional microphone on horizontal localization performance in hearing aid wearers

Abstract: This study examined the effect that signal processing strategies used in modern hearing aids, such as multi-channel WDRC, noise reduction, and directional microphones have on interaural difference cues and horizontal localization performance relative to linear, time-invariant amplification. Twelve participants were bilaterally fitted with BTE devices. Horizontal localization testing using a 360 degrees loudspeaker array and broadband pulsed pink noise was performed two weeks, and two months, post-fitting. The effect of noise reduction was measured with a constant noise present at 80 degrees azimuth. Data were analysed independently in the left/right and front/back dimension and showed that of the three signal processing strategies, directional microphones had the most significant effect on horizontal localization performance and over time. Specifically, a cardioid microphone could decrease front/back errors over time, whereas left/right errors increased when different microphones were fitted to left and right ears. Front/back confusions were generally prominent. Objective measurements of interaural differences on KEMAR explained significant shifts in left/right errors. In conclusion, there is scope for improving the sense of localization in hearing aid users.

Sound source separation device, speech recognition device, mobile telephone, sound source separation method, and program

Abstract: A sound source signal from a target sound source is allowed to be separated from a mixed sound which consists of sound source signals emitted from a plurality of sound sources without being affected by uneven sensitivity of microphone elements. A beamformer section 3 of a source separation device 1 performs beamforming processing for attenuating sound source signals arriving from directions symmetrical with respect to a perpendicular line to a straight line connecting two microphones 10 and 11 respectively by multiplying output signals from the microphones 10 and 11 after spectrum analysis by weighted coefficients which are complex conjugate to each other. Power computation sections 40 and 41 compute power spectrum information, and target sound spectrum extraction sections 50 and 51 extract spectrum information of a target sound source based on a difference between the power spectrum information.

A MEMS Condenser Microphone for Consumer Applications

Abstract: This paper describes the design, fabrication, testing, and characterization of a MEMS microphone fabricated at Analog Devices. The device consists of a polysilicon diaphragm suspended over a single crystal silicon backplate fabricated on silicon on insulator (SOI) wafers. The MEMS microphone has been successfully fabricated and tested in an anechoic chamber. The microphone is fabricated using a process that is compatible with inexpensive high volume production using unit processes that are currently used to fabricate inertial sensors. Details of the design, fabrication, and electrical and acoustic characterization of the microphone will be presented.

Methods and apparatus for targeted sound detection

Abstract: Targeted sound detection methods and apparatus are disclosed. A microphone array has two or more microphones M0 . . . MM. Each microphone is coupled to a plurality of filters. The filters are configured to filter input signals corresponding to sounds detected by the microphones thereby generating a filtered output. One or more sets of filter parameters for the plurality of filters are pre-calibrated to determine one or more corresponding pre-calibrated listening zones. Each set of filter parameters is selected to detect portions of the input signals corresponding to sounds originating within a given listening zone and filter out sounds originating outside the given listening zone. A particular pre-calibrated listening zone is selected at a runtime by applying to the plurality of filters a set of filter coefficients corresponding to the particular pre-calibrated listening zone. As a result, the microphone array may detect sounds originating within the particular listening sector and filter out sounds originating outside the particular listening zone.

Superdirective Beamforming Robust Against Microphone Mismatch

Abstract: Fixed superdirective beamformers using small-size microphone arrays are known to be highly sensitive to errors in the assumed microphone array characteristics. This paper discusses the design of robust superdirective beamformers by taking into account the statistics of the microphone characteristics. Different design procedures are considered: applying a white noise gain constraint, trading off the mean noise and distortion energy, and maximizing the mean or the minimum directivity factor. When computational complexity is not important, maximizing the mean or the minimum directivity factor is the preferred design procedure. In addition, it is shown how to determine a suitable parameter range for the other design procedures.

Security system and method for use of same

Abstract: A security system and method for use of the same are disclosed that provide for remote surveillance In one embodiment, a property's entry point such as a doorway is equipped with a video camera, an external microphone, and an external speaker An individual, such as an owner of the property, is away from the property and equipped with a cellular telephone When a person arrives at the entry point, a control unit relays audio and visual data captured by the video camera and the external microphone to the individual's cellular telephone Similarly, the control unit relays audio data from the owner to the person at the entry point

Voice control welding/cutting helmet functions and settings

Abstract: Provided for is a control system for a welding helmet comprising: an electronically controllable lens configured to be mounted in a welding helmet shell, a microphone configured to receive an audible input and to generate a signal in response to the audible input received and an electronic control module coupled to the lens and to the microphone and configured to control the electronically controllable lens based upon the signal. Also provided for is a welding helmet implementing a control system and a method of manufacturing a welding helmet including a control system.

The Development of a Biologically-Inspired Directional Microphone for Hearing Aids

Abstract: The development of novel microfabrication techniques for producing a directional microphone for hearing aids is here described. The mechanisms underlying both the structure and function of these unusu

Electrical Module Comprising a Mems Microphone

Abstract: An electrical module includes a base plate having an acoustic channel that opens into a first cavity at a first end and that is closed off by a microphone chip at a second end. The microphone chip borders a second cavity that opens to an exterior of the electrical module. The second cavity is separated from the acoustic channel by the microphone chip.

Wireless event authentication system

Abstract: A wireless data acquisition and recording (DAR) system includes acquisition circuitry including a video camera (18) and/or a microphone (20) carried by an operator for capturing video and audio events viewed and/or heard by the operator. A wireless transmitter (82) wirelessly transmits digital signals containing the video events captured by the video camera and the audio events detected by the microphone. A data display and report submission (DD) device (26) receives the wirelessly transmitted digital data captured by the acquisition circuitry and stores the data as media files.