University of Illinois

Motorola Corporation

Cypress Semiconductor Corporation

Other

Motorola Mobility LLC

Spansion LLC.

Chen Liu

Albert S Feng

William D O'Brien

Bruce C Wheeler

Charissa R Lansing

Douglas L Jones

Robert C Bilger

Richard M Fastow

Changxue Ma

Jens Olson

Yuan-Jun Wei

Stephan Rosner

Steven J Nowlan

Yan Ming Cheng

James R Talley

Ojas Ashok Bapat

Mark E Elledge

Michael E Lockwood

Jim Talley

Ted Mazurkiewicz

Carolyn T Bilger, Legal Representative

Lynette Melnar

Embodiments include a method, apparatus, and a computer program product for combining results from multiple decoders. For example, the method can include generating a network of paths based on one or more outputs associated with each of the multiple decoders. The network of paths can be scored to find an initial path with the highest path score based on scores associated with the one or more outputs. A weighting factor can be calculated for each of the multiple decoders based on a number of outputs from each of the multiple decoders included in the initial path with the highest path score. Further, the network of paths can be re-scored to find a new path with the highest path score based on the scores associated with the one or more outputs and the weighting factor for each of the multiple decoders.

Combining of results from multiple decoders

A voice activation system is provided. The voice activation system includes a first stage configured to output a first activation signal if at least one energy characteristic of a received audio signal satisfies at least one threshold and a second stage configured to transition from a first state to a second state in response to the first activation signal and, when in the second state, to output a second activation signal if at least a portion of a profile of the audio signal substantially matches at least one predetermined profile.

Power-efficient voice activation

Computer-based speech recognition can be improved by recognizing words with an accurate accent model. In order to provide a large number of possible accents, while providing real-time speech recognition, a language tree data structure of possible accents is provided in one embodiment such that a computerized speech recognition system can benefit from choosing among accent categories when searching for an appropriate accent model for speech recognition.

Recognition of speech with different accents

A communication device includes: (1) a wireless adapter at which a wireless headset is communicatively connected to the communication device and at which is received a first acoustic input that includes a speech input and a first ambient noise input; (2) a microphone that receives a second acoustic input, which includes a second ambient noise input; and (3) a dual-channel adaptive noise canceller that utilizes the second ambient noise input to filter the first ambient noise input out of the first acoustic input to generate an acoustic output that primarily comprises the speech input.

Noise reduction on wireless headset input via dual channel calibration within mobile phone

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

Acoustic phones (preferably drawnfrom a plurality of spoken languages) are providedA hierarchically-organized polyphone network () organizes views of these phones of varying resolution and phone categorization as a function, at least in part, of phonetic similarity () and at least one language-independent phonological factor (). In a preferred approach, a unique transcription system serves to represent the phones using only standard, printable ASCII characters, none of which comprises a special character (such as those characters that have a command significance for common script interpreters such as the UNIX command line).

Polyphone network method and apparatus

An electronic device () for speech dialog includes functions that receive () a speech phrase that comprises a request phrase that includes an instantiated variable (), generate () pitch and voicing characteristics () of the instantiated variable, and performs speech recognition () of the instantiated variable to determine a most likely set of acoustic states (). The electronic device may generate () a synthesized value of the instantiated variable using the most likely set of acoustic states and the pitch and voicing characteristics of the instantiated variable. The electronic device may use a table of previously entered values of variables that have been determined to be unique, and in which the values are associated with a most likely set of acoustic states and the pitch and voicing characteristics determined at the receipt of each value to disambiguate () a newly received instantiated variable.

Speech dialog method and system

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

A desired acoustic signal is extracted from a noisy environment by generating a signal representative of the desired signal with a processor for a hearing aid device. The processor receives binaural signals from two microphones at different locations. The binaural 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 position relative to a desired signal source. Location of the noise source is determined and the spectral content of the desired signal is determined from the intermediate signal corresponding to the noise source location. Inverse transformation of the selected intermediate signal followed by digital to analog conversion provides an output signal representative of the desired signal.

Growing community of inventors

Woodridge, IL, United States of America

Chen Liu