University of Illinois

Other

Teradyne, Inc.

Phonak Ag

Douglas L Jones

Albert S Feng

Bruce C Wheeler

William D O'Brien

Charissa R Lansing

Robert C Bilger

Chen Liu

Michael E Lockwood

Carolyn J Bilger, Legal Representative

Francois Callias

Billy J Hunsinger

Kannan Ramchandran

Mark E Elledge

Matthew D Kleffner

Scott D Jezwinski

Carolyn T Bilger, Legal Representative

Brian Scott Krongold

Xie Qi

Christopher D Schmitz

Jeffery B Larsen

Steve Franke

Mitesh Parikh

Crista Malick

Erik Larsen

François Callias

Qi Xie

A maximum-kurtosis, distortionless response (MKDR) technique and an extension, the maximum-kurtosis, Wiener estimate (MKWE) technique, are provided. In one form, blind estimates of the speech source's channel response are made from the microphone data and MVDR is applied. The source direction is estimated by finding weights that maximize output kurtosis, or the fourth central statistical moment, in the frequency domain. The MKWE approach approximates the Wiener filter by using MKDR-output noise power estimates to compute a Wiener post-filter. These approaches can be extended to block-adaptive versions if the speech source is not quickly moving in space.

Method and apparatus for blind signal recovery in noisy, reverberant environments

One embodiment of the invention includes a processing subsystem () responsive to the detection of sound coming from a room () to establish an acoustic impulse response for the room (), and determine a number of sound reflections corresponding to the impulse response. This subsystem () is further operable to determine volume of the room (), one or more dimensions of the room () and/or at least one absorption coefficient of a room boundary as a function of the sound reflections.

Room volume and room dimension estimation

System () is disclosed including an acoustic sensor array () coupled to processor (). System () processes inputs from array () to extract a desired acoustic signal through the suppression of interfering signals. The extraction/suppression is performed by modifying the array () inputs in the frequency domain with weights selected to minimize variance of the resulting output signal while maintaining unity gain of signals received in the direction of the desired acoustic signal. System () may be utilized in hearing aids, voice input devices, surveillance devices, and other applications.

Interference suppression techniques

System () is disclosed including an acoustic sensor array () coupled to processor (). System () processes inputs from array () to extract a desired acoustic signal through the suppression of interfering signals. The extraction/suppression is performed by modifying the array () inputs in the frequency domain with weights selected to minimize variance of the resulting output signal while maintaining unity gain of signals received in the direction of the desired acoustic signal. System () may be utilized in hearing, cochlear implants, speech recognition, voice input devices, surveillance devices, hands-free telephony devices, remote telepresence or teleconferencing, wireless acoustic sensor arrays, and other applications.

Systems and methods for interference suppression with directional sensing patterns

A number of ear-worn hearing system devices are provided that each include a pair of electrodes to transmit time varying electrical signals therebetween when in contact with skin of a user's body. The devices each include a housing, a sound sensor, and processing circuitry included within the housing. The electrodes are coupled to the circuitry and are spaced apart from one another a distance sufficient to provide capacitance between the electrodes below a desired threshold. The electrodes are disposed along the housing for placement on locations of the user's body where skin contact is not likely to be disrupted by nominal body movements.

Electrode placement for wireless intrabody communication between components of a hearing system

Disclosed is a hearing aid that includes a first component operable to receive an input and a second component operable to provide an output to at least one ear of a user of the hearing aid. The first component and the second component are electrically coupled together by a closed electrical circuit including at least one electrical conduction pathway formed by skin of the user between the first component and the second component. An electrical signal is transmitted from the first component to the second component with the closed electrical circuit. Communication between the first and second components can be one-way or two-way.

Intrabody communication for a hearing aid

Systems and methods for interference-suppression with directional sensing patterns

A desired acoustic signal is extracted from a noisy environment by generating a signal representative of the desired signal with a processor. The processor receives aural signals from two sensors each at a different location. The two inputs to the processor are converted from analog to digital format and then submitted to a discrete Fourier transform process to generate discrete spectral signal representations. The spectral signals are delayed by a number of time intervals in a dual delay line to provide a number of intermediate signals, each corresponding to a different spatial location relative to the two sensors. Locations of the noise source and the desired source are determined and the spectral content of the desired signal is determined from the intermediate signal corresponding to the noise source locations. Inverse transformation of the selected intermediate signal followed by digital to analog conversion provides an output signal representative of the desired signal. Techniques to localize multiple acoustic sources are also disclosed. Further, a technique to enhance noise reduction from multiple sources based on two-sensor reception is described.

Binaural signal processing techniques

A desired acoustic signal is extracted from a noisy environment by generating a signal representative of the desired signal with processor (). Processor () receives aural signals from two sensors () each at a different location. The two inputs to processor () are converted from analog to digital format and then submitted to a discrete Fourier transform process to generate discrete spectral signal representations. The spectral signals are delayed to provide a number of intermediate signals, each corresponding to a different spatial location relative to the two sensors. Locations of the noise source and the desired source, and the spectral content of the desired signal are determined from the intermediate signal corresponding to the noise source locations. Inverse transformation of the selected intermediate signal followed by digital to analog conversion provides an output signal representative of the desired signal with output device (). Techniques to localize multiple acoustic sources are also disclosed. Further, a technique to enhance noise reduction from multiple sources based on two-sensor reception is described.

Binaural signal processing using multiple acoustic sensors and digital filtering

Optimum bit rates and power levels are determined for subchannels in multicarrier communication systems. A tangent of the rate-power curve has a slope &lgr;. The slope is defined by the quotient difference between high/low power and high/low rate. A particular &lgr; is evaluated to find its corresponding total power followed by an update of &lgr;, in the form of an increase or decrease, to get closer to the optimal solution. Each &lgr; is evaluated to find the optimal operating point for each subchannel on the rate-power curve by summing the power allocated to the subchannels, and comparing the result to the power budget. Look-up tables are stored for individual channels, but similarity between channels permits joint use of look-up tables by multiple channels. The tables are used to determine the rate-power characteristics at each iteration. An optimal solution is found when either a newly chosen power allocation meets the power budget exactly or a newly chosen power budget equals the high or low power of a previous iteration.

Section division operating point determination method for multicarrier communication systems

An optical fiber test unit and a method of operating it. The test unit has a laser with a computer controlled excitation current input. To take a measurement, the laser is turned on for a period of time which is approximately equal to twice the propagation time through the fiber. The laser is then turned off and the reflected signal is recorded. This signal is displayed and contains discontinuities which indicate discontinuities in the fiber. The unit provides good dynamic range, but is able to make measurements more rapidly than a conventional OTDR. It is therefore well suited for scan testing of fiber optic cable bundles.

Method and apparatus for testing fiber optic telephone lines

Method and apparatus for adaptive filtering a first signal to produce a second signal that approximates a desired signal including an acoustic charge transport type programmable transversal filter (PTF) having a number of taps where each tap corresponds to an associated delayed first signal and each tap having means for providing a variable tap weight to modify the filter response. The first signal is coupled to the PTF and the output of the PTF produces the second signal. A summer compares the output of the PTF and the desired signal to produce an error signal. A programmable delay line has time delays programmable to integer multiples of the PTF tap spacing to provide a delayed version of the first signal, and a multiplier multiplies the error signal and the delayed input of the programmable delay line to produce a gradient signal. The gradient signal is filtered with an analog lowpass filter to produce a filtered gradient signal which in turn is converted to a digital signal by a low bandwidth analog to digital converter. A microprocessor implements a filtered gradient algorithm to update the PTF filter tap, load the new tap weight into the PTF, and reconfigure the programmable delay line to produce a filtered gradient for another tap. The comparison of the PTF output with the desired signal continues to develop an error signal and the adaptive process is repeated until the error signal becomes substantially zero which results in the second signal more closely approximating the desired signal.

Growing community of inventors

Champaign, IL, United States of America

Douglas L Jones