Brigham Young University

Sonic Innovations, Inc.

Other

Unisys Corporation

Sonix Technologies, Inc.

Douglas M Chabries

Richard Wesley Christiansen

Keith L Davis

Thomas G Stockham, Jr

Carver A Mead

Xiaoling Fang

Gary Robert Kenworthy

Darrell Rose

Robert N Brey

William Charles Borough

Martin Robinette

Aaron Michael Hammond

Richard L Frost

Christopher F Barnes

Douglas Lynn

A hearing compensation system comprises a plurality of bandpass filters having an input connected to an input transducer and each bandpass filter having an output connected to the input of one of a plurality of multiplicative automatic gain control (MAGC) circuits whose outputs are summed together and connected to the input of an output transducer. The MAGC circuits attenuate acoustic signals having a constant background level without the loss of speech intelligibility. The identification of the background noise portion of the acoustic signal is made by the constancy of the envelope of the input signal in each of the several frequency bands. The background noise that will be suppressed includes multi-talker speech babble, fan noise, feedback whistle, florescent light hum, and white noise. For use in the consumer electronics field background acoustic noise may be sensed and used to adjust gain in the various MAGC circuits so as to improve a user's listening experience, whether the user is hearing impaired or not.

Hearing compensation system incorporating signal processing techniques

A hearing compensation system for the hearing impaired comprises a plurality of bandpass filters having an input connected to an input transducer and each bandpass filter having an output connected to the input of one of a plurality of multiplicative AGC circuits whose outputs are summed together and connected to the input of an output transducer. The multiplicative AGC circuits attenuate acoustic signals having a constant background level without the loss of speech intelligibility. The identification of the background noise portion of the acoustic signal is made by the constancy of the envelope of the input signal in each of the several frequency bands. The background noise that will be suppressed includes multi-talker speech babble, fan noise, feedback whistle, florescent light hum, and white noise.

Hearing aid device incorporating signal processing techniques

A method for fitting a hearing compensation device comprises selecting a plurality of loudness levels for a plurality of frequencies and comparing each loudness level for each frequency for perceived sameness. The loudness levels may then be adjusted as needed to achieve perceived sameness across the frequency spectrum. A gain curve for each frequency is calculated from the selected plurality of loudness levels.

Hearing aid fitting system

A hearing compensation system for the hearing impaired comprises an input transducer for converting acoustical information at an input to electrical signals at an output, an output transducer for converting electrical signals at an input to acoustical information at an output, a plurality of bandpass filters, each bandpass filter having an input connected to the output of said input transducer, a plurality of AGC circuits, each individual AGC circuit associated with a different one of the bandpass filters and having an input connected to the output of its associated bandpass filter and an output connected to the input of the output transducer. The bandpass filters and AGC circuits may be divided into two processing channels, one for low frequencies and one for high frequencies and may drive separate audio transducers, one configured for maximum efficiency at low frequencies and one configured for maximum efficiency at high frequencies.

A hearing compensation system comprises an input transducer for converting acoustical information at an input thereof to electrical signals at an output thereof, a differential analog-to-digital converter sampling the electrical signals output from the input transducer at an input thereof and outputting differential signal samples at an output thereof, a digital signal processing circuit having an input connected to the output of the differential analog-to-digital converter and operating on the differential signal samples to form processed differential signal samples at an output thereof, and an output transducer for converting electrical signals at an input thereof to acoustical information at an output thereof, the processed differential signal samples coupled to the input of the output transducer.

Digital hearing aid using differential signal representations

A hearing compensation system for the hearing impaired comprises an input transducer for converting acoustical information at an input to electrical signals at an output, an output transducer for converting electrical signals at an input to acoustical information at an output, a plurality of bandpass filters, each bandpass filter having an input connected to the output of said input transducer, a plurality of AGC circuits, each individual AGC circuit associated with a different one of the bandpass filters and having an input connected to the output of its associated bandpass filter and an output connected to the input of the output transducer.

A multistage data compression system includes an encoder for receiving input data sequences from a data source, where the encoder comprises two or more quantizer stages, each stage having a memory for storing an established set of codewords preselected such that the sum of different combinations of codewords, one for each stage, is representative of likely to occur input data sequences. Each codeword of each stage has a respective index associated therewith. The encoder operates to select a codeword from at least two of the stages such that the sum of the selected codewords is a closer match to each input sequence than any other sum of codewords formed from the selected stages. When the codewords of interest are determined, the indices of those codewords are retrieved. These indices may then be transmitted to another location, stored in memory or otherwise processed. The system also includes a decoder coupled to the transmission channel, memory, or other mechanism which operates on the indices. The decoder includes a decoder memory for storing the established sets of codewords with respective indices, a retrieval device for retrieving from the decoder memory the codewords whose respective indices were transmitted, stored or otherwise processed, and a summing mechanism for adding the retrieved codewords to produce a reproduction data sequence representing an approximation of a corresponding input data sequence.

System and method for data compression using multiple codewords and

An apparatus and technique for enhancing the hearing capabilities of persons by providing a device which is a model of the desired hearing characteristic of the persons.

Digital hearing enhancement apparatus

An electronic system for processing sampled data input signals includes an electronic memory which stores a set of preprocessed vectors V.sub.1 f(m, . . . ) thru V.sub.N *f(m, . . . ) where f(m, . . . ) is a sampled data function, having any number of dimensions m, . . . . * is a convolution operator, and V.sub.1 thru V.sub.N are a finite set of N unprocessed vectors each of which represents an anticipated group of input signal samples. After these preprocessed vectors are stored, input signals are processed in real time by (a) a circuit which samples the input signal; (b) a circuit which compresses the sequence of samples that is taken into a smaller sequence of index signals that correspond to the indexes 1 thru N of the unprocessed vectors; (c) a circuit which receives the smaller sequence of index signals and reads from the memory those preprocessed vectors whose indexes match the received index signals; and (d) a circuit which adds together the read preprocessed vectors while maintaining a predetermined offset between them as they are added.

Method and apparatus for processing sampled data signals by utilizing

Synthetic aperture radar (SAR) images are formed by storing in electronic memory possible outputs of a lossy data compressor, f.sub.1,1 [m], f.sub.1,2 [m], . . . f.sub.1,P [m], where f.sub.1,i [m] represents a segment of the unprocessed radar return; by storing range-correlated segments of the SAR data, g.sub.1,1 [m], g.sub.1,2 [m], . . . g.sub.1,P [m], where g.sub.1,i [m] is the range-correlated version of f.sub.1,i [m]; by storing all possible outputs of a second lossy data compressor, f.sub.2,1 [n], f.sub.2,2 [n], . . . f.sub.2,Q [n], where f.sub.2,i [n] represents a segment of the range-correlated SAR image; and by storing segments of the range-azimuth focussed SAR image data, g.sub.2,1,m [n],g.sub.2,2,m [n], . . . g.sub.2,Q,m [n], where g.sub.2,i,m [n] represents a segment of the SAR range-azimuth focussed image. Thereafter, a sequence of separate radar pulses are transmitted from an aircraft and the return signal is sampled and placed in an array. Vectors of the array are compared with segments f.sub.1,i [m], and the indices of the segments which most closely match the vectors are stored in an array. An intermediate image is formed by adding together segments, g.sub.1,i [m] whose indices match those in the array, while maintaining a predetermined offset. Vectors of this intermediate image are compared with segments f.sub.2,i [n] and the indices of these segments which most closely match the intermediate range vectors are stored in a second array. The final image is formed by adding together segments, g.sub.2,i,m [n], whose indices match those in the second array, while maintaining a predetermined offset between each segment during the addition.

Data compression method and apparatus for radar image formation and like

An adaptive noise suppressor for providing noise filtered signals. The noise suppression device employs a vector gain .mu. for the weights of the filter wherein the vector .mu. is selected for each frequency bin to be inversely proportional to the power spectrum. A projection operator is utilized to remove the effects of circular convolution to produce a linear convolution result wherein the weights are readjusted in a manner to minimize the difference between the input signal and the filter output signal, thereby minimizing the error signal to produce a noise suppressed signal in the filtered output. A frequency suppression device utilizes the same principles of the vector .mu. and projection operator, but the output is taken from the error output of the filter.

Growing community of inventors

Woodland Hills, UT, United States of America

Douglas M Chabries