Abstract: A system for detection of at least one designated wake-up word for at least one speech-enabled application. The system comprises at least one microphone; and at least one computer hardware processor configured to perform: receiving an acoustic signal generated by the at least one microphone at least in part as a result of receiving an utterance spoken by a speaker; obtaining information indicative of the speaker's identity; interpreting the acoustic signal at least in part by determining, using the information indicative of the speaker's identity and automated speech recognition, whether the utterance spoken by the speaker includes the at least one designated wake-up word; and interacting with the speaker based, at least in part, on results of the interpreting.

Abstract: A system for detection of at least one designated wake-up word for at least one speech-enabled application. The system comprises at least one microphone; and at least one computer hardware processor configured to perform: receiving an acoustic signal generated by the at least one microphone at least in part as a result of receiving an utterance spoken by a speaker; obtaining information indicative of the speaker's identity; interpreting the acoustic signal at least in part by determining, using the information indicative of the speaker's identity and automated speech recognition, whether the utterance spoken by the speaker includes the at least one designated wake-up word; and interacting with the speaker based, at least in part, on results of the interpreting.

Abstract: There is provided a speech dialog system that includes a first microphone, a second microphone, a processor and a memory. The first microphone captures first audio from a first spatial zone, and produces a first audio signal. The second microphone captures second audio from a second spatial zone, and produces a second audio signal. The processor receives the first audio signal and the second audio signal, and the memory contains instructions that control the processor to perform operations of a speech enhancement module, an automatic speech recognition module, and a speech dialog module that performs a zone-dedicated speech dialog.

Abstract: Gain mismatch and related problems can be solved by a system and method that applies an automatic microphone signal gain equalization without any direct absolute reference or calibration phase. The system and method performs the steps of receiving, by a computing device, a speech signal from a speaking person via a plurality of microphones, determining a speech signal component in the time-frequency domain for each microphone of the plurality of microphones, calculating an instantaneous cross-talk coupling matrix based on the speech signal components across the microphones, estimating gain factors based on calculated cross-talk couplings and a given expected cross-talk attenuation, limiting the gain factors to appropriate maximum and minimum values, and applying the gain factors to the speech signal used in the control path to control further speech enhancement algorithms or used in the signal path for direct influence on the speech enhanced audio output signal.

Abstract: There is provided a speech dialog system that includes a first microphone, a second microphone, a processor and a memory. The first microphone captures first audio from a first spatial zone, and produces a first audio signal. The second microphone captures second audio from a second spatial zone, and produces a second audio signal. The processor receives the first audio signal and the second audio signal, and the memory contains instructions that control the processor to perform operations of a speech enhancement module, an automatic speech recognition module, and a speech dialog module that performs a zone-dedicated speech dialog.

Abstract: Gain mismatch and related problems can be solved by a system and method that applies an automatic microphone signal gain equalization without any direct absolute reference or calibration phase. The system and method performs the steps of receiving, by a computing device, a speech signal from a speaking person via a plurality of microphones, determining a speech signal component in the time-frequency domain for each microphone of the plurality of microphones, calculating an instantaneous cross-talk coupling matrix based on the speech signal components across the microphones, estimating gain factors based on calculated cross-talk couplings and a given expected cross-talk attenuation, limiting the gain factors to appropriate maximum and minimum values, and applying the gain factors to the speech signal used in the control path to control further speech enhancement algorithms or used in the signal path for direct influence on the speech enhanced audio output signal.

Abstract: A method, computer program product, and computer system for identifying, by a computing device, a plurality of sources, wherein a first source of the plurality of sources is a source of interest and wherein a second source of the plurality of sources is an interference source. The first source and the second source may be monitored simultaneously by implementing a spatial pre-filter for acoustic source localization.

Abstract: Methods and apparatus for frequency selective signal mixing for speech enhancement. In one embodiment frequency-based channel selection is performed for signal magnitude, signal energy, and noise estimate using speaker activity detection information, signal-to-noise ratio, and/or signal level, Frequency-based channel selection is performed for a dynamic spectral floor to adjust the noise estimate using speaker dominance information. Noise reduction is performed on the signal for the selected channel.

Abstract: A system for detection of at least one designated wake-up word for at least one speech-enabled application. The system comprises at least one microphone; and at least one computer hardware processor configured to perform: receiving an acoustic signal generated by the at least one microphone at least in part as a result of receiving an utterance spoken by a speaker; obtaining information indicative of the speaker's identity; interpreting the acoustic signal at least in part by determining, using the information indicative of the speaker's identity and automated speech recognition, whether the utterance spoken by the speaker includes the at least one designated wake-up word; and interacting with the speaker based, at least in part, on results of the interpreting.

Abstract: A method, computer program product, and computer system for receiving, by a computing device, a speech signal from a speaker via a plurality of microphone zones. A temporal cue based confidence may be determined for at least a portion of the plurality of microphone zones. A power cue based confidence may be determined for at least a portion of the plurality of microphone zones. A microphone zone of the plurality of microphone zones from which to use an output signal of the speaker may be identified based upon, at least in part, a combination of the temporal cue based confidence and the power cue based confidence.

Abstract: A method, computer program product, and computer system for receiving, by a computing device, a speech signal from a speaker via a plurality of microphone zones. A temporal cue based confidence may be determined for at least a portion of the plurality of microphone zones. A power cue based confidence may be determined for at least a portion of the plurality of microphone zones. A microphone zone of the plurality of microphone zones from which to use an output signal of the speaker may be identified based upon, at least in part, a combination of the temporal cue based confidence and the power cue based confidence.

Abstract: Embodiments of the present disclosure may include a system and method for speech enhancement using the coherent to diffuse sound ratio. Embodiments may include receiving an audio signal at one or more microphones and controlling one or more adaptive filters of a beamformer using a coherent to diffuse ratio (“CDR”).

Abstract: Embodiments disclosed herein may include determining a signal parameter of a first microphone and a second microphone associated with a computing device. Embodiments may include generating a reference parameter based upon at least one of the parameter of the first microphone and the parameter of the second microphone. Embodiments may include adjusting a tolerance of at least one of the first microphone and the second microphone, based upon the reference parameter. Embodiments may include receiving, at the first microphone, a first speech signal, the first speech signal having a first speech signal magnitude and receiving, at the second microphone, a second speech signal, the second speech signal having a second speech signal magnitude. Embodiments may include comparing at least one of the first speech signal magnitude and the second speech signal magnitude with a third speech signal magnitude and detecting an obstructed microphone based upon the comparison.

Abstract: A multi-mode speech communication system is described that has different operating modes for different speech applications. A signal processing module is in communication with the speech applications and includes an input processing module and an output processing module. The input processing module processes microphone input signals to produce a set user input signals for each speech application that are limited to currently active system users for that speech application. The output processing module processes application output communications from the speech applications to produce loudspeaker output signals to the system users, wherein for each different speech application, the loudspeaker output signals are directed only to system users currently active in that speech application.

Abstract: Embodiments disclosed herein may include determining a signal parameter of a first microphone and a second microphone associated with a computing device. Embodiments may include generating a reference parameter based upon at least one of the parameter of the first microphone and the parameter of the second microphone. Embodiments may include adjusting a tolerance of at least one of the first microphone and the second microphone, based upon the reference parameter. Embodiments may include receiving, at the first microphone, a first speech signal, the first speech signal having a first speech signal magnitude and receiving, at the second microphone, a second speech signal, the second speech signal having a second speech signal magnitude. Embodiments may include comparing at least one of the first speech signal magnitude and the second speech signal magnitude with a third speech signal magnitude and detecting an obstructed microphone based upon the comparison.

Abstract: Embodiments of the present disclosure may include a system and method for speech enhancement using the coherent to diffuse sound ratio.

Abstract: Method and apparatus to determine a speaker activity detection measure from energy-based characteristics of signals from a plurality of speaker-dedicated microphones, detect acoustic events using power spectra for the microphone signals, and determine a robust speaker activity detection measure from the speaker activity measure and the detected acoustic events.

Abstract: A multi-mode speech communication system is described that has different operating modes for different speech applications. A signal processing module is in communication with the speech applications and includes an input processing module and an output processing module. The input processing module processes microphone input signals to produce a set user input signals for each speech application that are limited to currently active system users for that speech application. The output processing module processes application output communications from the speech applications to produce loudspeaker output signals to the system users, wherein for each different speech application, the loudspeaker output signals are directed only to system users currently active in that speech application.

Abstract: A multi-mode speech communication system is described that has different operating modes for different speech applications. A speech service compartment contains multiple system users, multiple input microphones that develop microphone input signals from the system users to the system, and multiple output loudspeakers that develop loudspeaker output signals from the system to the system users. A signal processing module is in communication with the speech applications and includes an input processing module and an output processing module. The input processing module processes the microphone input signals to produce a set user input signals for each speech application that are limited to currently active system users for that speech application.

Abstract: Methods and apparatus for frequency selective signal mixing for speech enhancement. In one embodiment frequency-based channel selection is performed for signal magnitude, signal energy, and noise estimate using speaker activity detection information, signal-to-noise ratio, and/or signal level, Frequency-based channel selection is performed for a dynamic spectral floor to adjust the noise estimate using speaker dominance information. Noise reduction is performed on the signal for the selected channel.