Showing papers on "Microphone published in 2001"

Microphone Arrays Signal Processing Techniques and Applications

Abstract: I. Speech Enhancement.- 1 Constant Directivity Beamforming.- 2 Superdirective Microphone Arrays.- 3 Post-Filtering Techniques.- 4 Spatial Coherence Functions for Differential Microphones in Isotropic Noise Fields.- 5 Robust Adaptive Beamforming.- 6 GSVD-Based Optimal Filtering for Multi-Microphone Speech Enhancement.- 7 Explicit Speech Modeling for Microphone Array Speech Acquisition.- II. Source Localization.- 8 Robust Localization in Reverberant Rooms.- 9 Multi-Source Localization Strategies.- 10 Joint Audio-Video Signal Processing for Object Localization and Tracking.- III. Applications.- 11 Microphone-Array Hearing Aids.- 12 Small Microphone Arrays with Postfilters for Noise and Acoustic Echo Reduction.- 13 Acoustic Echo Cancellation for Beamforming Microphone Arrays.- 14 Optimal and Adaptive Microphone Arrays for Speech Input in Automobiles.- 15 Speech Recognition with Microphone Arrays.- 16 Blind Separation of Acoustic Signals.- IV. Open Problems and Future Directions.- 17 Future Directions for Microphone Arrays.- 18 Future Directions in Microphone Array Processing.

Post-Filtering Techniques

Abstract: In the context of microphone arrays, the term post-filtering denotes the post-processing of the array output by a single-channel noise suppression filter. A theoretical analysis shows that Wiener post-filtering of the output of an optimum distortionless beamformer provides a minimum mean squared error solution. We examine published methods for post-filter estimation and develop a new algorithm. A simulation system is presented to compare the performance of the discussed algorithms.

Investigation of acoustic noise on 15 MRI scanners from 0.2 T to 3 T.

Abstract: Acoustic noise levels for fast MRI pulse sequences were surveyed on 14 systems with field strengths ranging from 0.2 T to 3 T. A microphone insensitive to the magnetic environment was placed close to the magnet isocenter and connected via an extension cable to a sound level meter outside the scan room. Measured noise levels varied from 82.5 ± 0.1 dB(A) for a 0.23 T system to 118.4 ± 1.3 dB(A) for a 3 T system. Further measurements on four of the closed-bore systems surveyed showed that: 1) pulse sequence parameters (particularly FOV and TR) were more influential in determining noise level than field strength, 2) the noise level was found to vary along the z-direction with a maximum near the bore entrance, and 3) in one of two systems tested there was a significant increase in noise with a volunteer present instead of a test object. The results underline the importance of hearing protection for patients and for staff spending extended periods in the scan room. J. Magn. Reson. Imaging 2001;13:288–293. © 2001 Wiley-Liss, Inc.

Speech dereverberation via maximum-kurtosis subband adaptive filtering

Abstract: This paper presents an efficient algorithm for high-quality speech capture in applications such as hands-free teleconferencing or voice recording by personal computers. We process the microphone signals by a subband adaptive filtering structure using a modulated complex lapped transform (MCLT), in which the subband filters are adapted to maximize the kurtosis of the linear prediction (LP) residual of the reconstructed speech. In this way, we attain good solutions to the problem of blind speech dereverberation. Experimental results with actual data, as well as with artificially difficult reverberant situations, show very good performance, both in terms of a significant reduction of the perceived reverberation, as well as improvement in spectral fidelity.

Sound-based proximity detector

Abstract: A proximity detector for use in a mobile telephone having at least a microphone and a loudspeaker operatively connected to a signal processor is presented. The proximity detector includes data processing and control modules having a module for controlling the signal processor for activating the loudspeaker to reproduce an acoustic control signal. A correlator correlates a control signal received directly by the microphone and a control signal being reflected from a user of the telephone and then received by the microphone to determine the distance between the telephone and the user. A signal level controller controls the signal processor to vary the signal level of an audible signal reproduced by the loudspeaker depending on the determined distance between the telephone and the user.

On-vehicle handsfree system and mobile terminal thereof

Abstract: A handsfree video phone system includes a video phone terminal and a car device in which a microphone, a speaker, and a camera are connected with an instrument panel via a cable. The video phone terminal is connected with the instrument panel via a wireless communication medium. When the video phone terminal is brought into the vehicle, the instrument panel informs the video phone terminal of the attribute information on the system structure. According to the attribute information, the video phone terminal switches the paths of multiplexed audio-video data so as to replace a microphone, a speaker, a camera, and a display that have been built in the video phone terminal with the microphone, the speaker, the camera, and a display of the instrument panel when the status changes to the communication status.

Active noise control for periodic disturbances

Abstract: Proposes an active noise control algorithm for periodic disturbances of unknown frequency. The algorithm is appropriate for the feedback case in which a single error microphone is used. A previously proposed algorithm for the rejection of sinusoidal noise sources is extended for the cancellation of multiple harmonics. Unlike many other approaches, the estimates of the frequencies of the separate harmonics are tied together within the algorithm to account for the integer multiplicative relations between them. The dynamic behavior of the closed-loop system is analyzed using an approximation that is shown, in simulations, to provide an accurate representation of the system's behavior. Experimental results on an active noise control testbed demonstrate the success of the method in a practical environment.

Microphone array apparatus

Abstract: A microphone array apparatus includes a microphone array including microphones, one of the microphones being a reference microphone, filters receiving output signals of the microphones, and a filter coefficient calculator which receives the output signals of the microphones, a noise and a residual signal obtained by subtracting filtered output signals of the microphones other than the reference microphone from a filtered output signal of the reference microphone and which obtain filter coefficients of the filters in accordance with an evaluation function based on the residual signal.

Auto-calibrating surround system

Abstract: A multi-channel surround sound system and method is described that allows automatic and independent calibration and adjustment of the frequency, amplitude and time response of each channel of the surround sound system. The disclosed auto-calibrating surround sound (ACSS) system includes a processor that generates a test signal represented by a temporal maximum length sequence (MLS) and supplies the test signal as part of an electric input signal to a loudspeaker. A microphone coupled to the processor receives the signal in a listening environment. The processor correlates the received sound signal with the test signal in the time domain and determines from the correlated signals a whitened response of the audio channel in the listening environment.

Method of providing a hydrophobic layer and a condenser microphone having such a layer

Abstract: A method of providing at least part of a diaphragm and at least a part of a back-plate of a condenser microphone with ahydrophobic layer so as to avoid stiction between said diaphragm and said back-plate. The layer is deposited via a number small of openings in the back-plate, the diaphragm and/or between the diaphragm and the back-plate. Provides a homogeneous and structured hydrophobic layer, even to small internal cavities of the microstructure. The layer may be deposited by a liquid phase or a vapour phase deposition method. The method may be applied naturally in continuation of the normal manufacturing process. Further, a MEMS condenser microphone is provided having such a hydrophobic layer. The static distance between the diaphragm and the back-plate of the microphone is smaller than 10 νm.

Speech recognition system

Abstract: A speech recognition system 7 for an automotive vehicle 10 is provided including a microphone receiver 12 for receiving a voice audio signal and converting the same to an analog signal 13, a microphone aimer for giving the microphone receiver 12 a locational bias for reception, an analog to digital converter 15 for converting the analog signal to a digital signal, a speech recognizer 17 for recognizing a voice from the digital signal received from the analog to digital converter 15, an occupant restraint system 22 having an occupant informational system 60/18 to control deployment of the occupant restraint system 22 resultant upon an occupant condition, an occupant restraint system signal generator 18 for signaling the occupant condition to the microphone aimer to locationally bias the reception of the microphone receiver 12.

CMOS-MEMS membrane for audio-frequency acoustic actuation

Abstract: Using CMOS-MEMS micromachining techniques we have constructed a prototype earphone that is audible from 1 to 15 kHz. The fabrication of the acoustic membrane consists of only two steps in addition to the prior post-CMOS micromachining steps developed at CMU. The ability to build a membrane directly on a standard CMOS chip, integrating mechanical structures with signal processing electronics will enable a variety of applications including economical earphones, microphones, hearing aids, high-fidelity earphones, cellular phones and noise cancellation. The large compliance of the CMOS-MEMS membrane also promises application as a sensitive microphone and pressure sensor.

Single chip device for voice communications

Abstract: The present invention is a chip for use in a voice communication device. The chip provides for both bone conduction sensing and air conduction sensing. The chip includes a bone conduction sensing pattern disposed within the chip and a microphone sensing pattern disposed within the chip. In addition, the chip can optionally include an integrated circuit portion interconnected to the bone conduction sensing pattern and the microphone sensing pattern. The pattern can be of a piezoelectric polymer, the patterns overlaying the substrate. Preferably, the bone conduction sensing pattern and the microphone sensing pattern are placed on opposite ends of the chip.

Directional microphone assembly

Abstract: A directional microphone assembly for a hearing aid is disclosed. The hearing aid has one or more microphone cartridge(s), and first and second sound passages. Inlets to the sound passages, or the sound passages themselves, are spaced apart such that the shortest distance between them is less than or approximately equal to the length of the microphone cartridge(s). A sound duct and at least one surface of a microphone cartridge may form each sound passage, where the sound duct is mounted with the microphone cartridge. Alternatively, each sound duct may be formed as an integral part of a microphone cartridge.

Speech recognition in noise for cochlear implantees with a two-microphone monaural adaptive noise reduction system

Abstract: ObjectiveIn this study the performance of a noise reduction strategy applied to cochlear implants is evaluated. The noise reduction strategy is based on a 2-channel adaptive filtering strategy using two microphones in a single behind-the-ear hearing aid.DesignFour adult LAURA cochlear implant users

Heart-sound analyzing apparatus

Abstract: A heart-sound analyzing apparatus, including a heart-sound microphone which is adapted to be worn on a living subject to iteratively detect a second heart sound II of the subject, a respiration-synchronous-signal detecting device which iteratively detects a respiration-synchronous signal of the subject, an inspiration-expiration judging device for judging, based on each of the respiration-synchronous signals iteratively detected by the respiration-synchronous-signal detecting device, whether the subject is in an inspiring state or in an expiring state, and an aortic-valve-closing-timing determining device for iteratively determining a timing when the aortic valve of the heart of the subject closes, based on each of second heart sounds II which are iteratively detected by the heart-sound microphone in time intervals, respectively, in each of which the subject is judged as being in the inspiring state by the inspiration-expiration judging device.

Robust Speaker Recognition Using Microphone Arrays

Abstract: This paper investigates the use of microphone arrays in handsfree speaker recognition systems. Hands-free operation is preferable in many potential speaker recognition applications, however obtaining acceptable performance with a single distant microphone is problematic in real noise conditions. A possible solution to this problem is the use of microphone arrays, which have the capacity to enhance a signal based purely on knowledge of its direction of arrival. The use of microphone arrays for improving the robustness of speech recognition systems has been studied in recent times, however little research has been conducted in the area of speaker recognition. This paper discusses the application of microphone arrays to speaker recognition applications, and presents an experimental evaluation of a hands-free speaker verification application in noisy conditions.

System and method for extracting, decoding, and utilizing hidden data embedded in audio signals

Abstract: A system and method for enabling a user to retrieve, decode, and utilize hidden data embedded in audio signals. An exemplary implementation includes a microphone structured to receive sound waves representative of an audio signal and hidden data embedded in the audio signal. The then microphone converts the received sound waves into an electrical output signal. The system also includes a processor electrically coupled to the microphone and configured to receive the electrical output signal in order to extract the hidden data and provide information represented by the hidden data as an output thereof. A user interface is also provided and is electrically coupled to the processor and configured to receiver a first input from the user and activate the processor to selectively initiate extraction of the hidden data. The processor produces as an output the information represented by the hidden data. Finally, the system includes a user presentation mechanism configured to present the information to the user.

Teleconferencing system which selects and optimal acoustic reception pattern from multiple patterns having different shapes and sizes

Abstract: A telephone system includes two or more cardioid microphones held together and directed outwardly from a central point. Mixing circuitry and control circuitry combines and analyzes signals from the microphones and selects the signal from one of the microphones or from one of one or more predetermined combinations of microphone signals in order to track a speaker as the speaker moves about a room or as various speakers situated about the room speak then fall silent. Visual indicators, in the form of light emitting diodes (LEDs) are evenly spaced around the perimeter of a circle concentric with the microphone array. Mixing circuitry produces ten combination signals, A+B, A+C, B+C, A+B+C, A-B, B-C, A-C, A-0.5(B+C), B-0.5(A+C), and C-0.5(B+A), with the "listening beam" formed by combinations, such as A-0.5(B+C), that involve the subtraction of signals, generally being more narrowly directed than beams formed by combinations, such as A+B, that involve only the addition of signals. An omnidirectional combination A+B+C is employed when active speakers are widely scattered throughout the room. Weighting factors are employed in a known manner to provide unity gain output. Control circuitry selects the signal from the microphone or from one of the predetermined microphone combinations, based generally on the energy level of the signal, and employs the selected signal as the output signal. The control circuitry also operates to limit dithering between microphones and, by analyzing the beam selection pattern, may switch to a broader coverage pattern, rather than switching between two narrower beams that each covers one of the speakers.

Speech recognition using microphone antenna array

Abstract: A system and method of audio processing provides enhanced speech recognition. Audio input is received at a plurality of microphones. The multi-channel audio signal from the microphones may be processed by a beamforming network to generate a single-channel enhanced audio signal, on which voice activity is detected. Audio signals from the microphones are additionally processed by an adaptable noise cancellation filter having variable filter coefficients to generate a noise-suppressed audio signal. The variable filter coefficients are updated during periods of voice inactivity. A speech recognition engine may apply a speech recognition algorithm to the noise-suppressed audio signal and generate an appropriate output. The operation of the speech recognition engine and the adaptable noise cancellation filter may advantageously be controlled based on voice activity detected in the single-channel enhanced audio signal from the beamforming network.

Method for acoustic transducer calibration

Abstract: A method of acoustic transducer calibration ( 200, 400 ) using a band limited pseudo random noise source with an internal digital signal processor ( 209, 403 ) to tailor audio characteristics of an internal microphone 103 and internal speaker ( 301 ) within a communications device ( 101 ) to insure consistent amplitude and frequency characteristics of these microphone and speaker transducer devices. The method offers and advantage such that tuning of the amplitude and frequency response consistently converges to the desired filter response with a filter type offering operational stability.

Car reverse alerting and multi-functional display

Abstract: A car reverse alerting and multi-functional display is disclosed. A display and trumpet are mounted on the multi-functional rear view mirror of a car. A charge coupling device, a microphone, and a car reverse sensor for monitoring the images, sounds and distance at a backside of the car are installed. The monitored analog electronic signals are decoded through a decoder so that the electronic signal is converted into a digital signal from an analog signal. Then, these signals are transferred to the multi-functional rear view mirror by cable or wireless devices. Then the signals are processed by a central processing unit. Then an image signal is outputted to a display for displaying; and meanwhile, the display outputs the sound signal from the microphone. The driver can reverse a car safely from the image and speech alert.

Method and apparatus for an adaptive binaural beamforming system

Abstract: An adaptive binaural beamforming system is provided which can be used, for example, in a hearing aid. The system uses more than two input signals, and preferably four input signals. The signals can be provided, for example, by two microphone pairs, one pair of microphones located in a user's left ear and the second pair of microphones located in the user's right ear. The system is preferably arranged such that each pair of microphones utilizes an end-fire configuration with the two pairs of microphones being combined in a broadside configuration. Signal processing is divided into two stages. In the first stage, the outputs from the two microphone pairs are processed utilizing an end-fire array processing scheme, this stage providing the benefits of spatial processing. In the second stage, the outputs from the two end-fire arrays are processed utilizing a broadside configuration, this stage providing further spatial processing benefits along with the benefits of binaural processing.

Microphone array system

Abstract: PROBLEM TO BE SOLVED: To estimate sound receiving signals at positions on a co-axis by estimating a sound receiving signal at an optional position on a co-axis with two microphones from sound receiving signals from each microphone so as to place the two microphones on one axis. SOLUTION: A sound receiving signal estimate processing section assumes a sound wave from a sound source reaching two microphones to be a plane wave, expresses an estimate sound receiving signal at a position on a straight line tying the two microphones by a wave equation shown in Expression, estimates a coefficient b cosθdepending on an incoming direction of a sound wave expressed in the wave equation expressed in the Expression by assuming the mean power of the sound waves arrived respectively in the two microphones to be equal to each other, and then estimates a sound receiving signal at an optional position on the co-axis of the microphones on the basis of the sound receiving signals from the two microphones. In the Expression, x, y, z are each space axis, t is a time, v is a velocity of air particles, p is a sound pressure, a, b are coefficients, and θ indicates a direction of the sound source. Thus, the sound receiving signal at an optional position on the co-axis can be estimated.

Wearable computer system

Abstract: This invention involves a wearable computer having computer components movably located in a collar that the user wears around his or her neck. The computer components can be a display or monitor, or a microphone or any other computer component.

Voice control system with multiple voice recognition engines

Abstract: Described is a control unit including a recognition result receiver able to receive a recognition result, a recognition result association unit having associations of results with recognition engines, and a recognition engine activator able to activate the recognition engine associated with the recognition result. Also described is a device including a microphone, an analog to digital converter able to convert input received by the microphone, a first speech recognition engine adapted to perform a first type of recognition on an output of the analog to digital converter, and a second recognition engine adapted to perform a second type of recognition on the output.

The first low voltage, low noise differential silicon microphone, technology development and measurement results

Abstract: The first differential silicon microphone is presented. This capacitive working device consists of two backplates with a membrane in between. Due to the balanced arrangement the air gap can be minimized. Thus, a higher electrical field and sensitivity can be achieved for low voltages. A dedicated process sequence has been developed in order to get the optimum mechanical and electrical properties for all structural layers. Furthermore, a sandwich structure has been developed to achieve a reproducible, very sensitive microphone membrane with a thickness of only 0.5 /spl mu/m and a stress of 45 MPa. The total sensitivity for a bias of 1.5 V was measured to be 13 mV/Pa and the A-weighted equivalent input noise was measured to be 22.5 dB SPLA. The upper limit of the dynamic range has been determined to be 118 dB SPL and the total harmonic distortion at 80 dB SPL is below 0.26%.

Adaptive microphone matching in multi-microphone directional system

Abstract: Improved approaches to matching sensitivities of microphones in multi-microphone directional processing systems. These approaches operate to adaptively match microphone sensitivities so that directional noise suppression is robust. As a result, microphone sensitivities remain matched not only over time but also while in actual use. These approaches are particularly useful for hearing aid applications in which directional noise suppression is important.

A two-microphone dual delay-line approach for extraction of a speech sound in the presence of multiple interferers

Abstract: This paper describes algorithms for signal extraction for use as a front-end of telecommunication devices, speech recognition systems, as well as hearing aids that operate in noisy environments. The development was based on some independent, hypothesized theories of the computational mechanics of biological systems in which directional hearing is enabled mainly by binaural processing of interaural directional cues. Our system uses two microphones as input devices and a signal processing method based on the two input channels. The signal processing procedure comprises two major stages: (i) source localization, and (ii) cancellation of noise sources based on knowledge of the locations of all sound sources. The source localization, detailed in our previous paper [Liu et al., J. Acoust. Soc. Am. 108, 1888 (2000)], was based on a well-recognized biological architecture comprising a dual delay-line and a coincidence detection mechanism. This paper focuses on description of the noise cancellation stage. We designed a simple subtraction method which, when strategically employed over the dual delay-line structure in the broadband manner, can effectively cancel multiple interfering sound sources and consequently enhance the desired signal. We obtained an 8–10 dB enhancement for the desired speech in the situations of four talkers in the anechoic acoustic test (or 7–10 dB enhancement in the situations of six talkers in the computer simulation) when all the sounds were equally intense and temporally aligned.