Abstract: A vehicle defines an interior space and an exterior space. Within the vehicle are internal microphones that are disposed to capture an acoustic event that originated in an origination space, which is either the interior space or the exterior space. An infotainment system includes circuitry that forms a head unit having an acoustic-signal processor that is configured to receive, from the microphones, a sound vector indicative of the acoustic event and to identify the origination space based at least in part on the sound vector.

Abstract: According to some aspects, a system for detecting a designated wake-up word is provided, the system comprising a plurality of microphones to detect acoustic information from a physical space having a plurality of acoustic zones, at least one processor configured to receive a first acoustic signal representing the acoustic information received by the plurality of microphones, process the first acoustic signal to identify content of the first acoustic signal originating from each of the plurality of acoustic zones, provide a plurality of second acoustic signals, each of the plurality of second acoustic signals substantially corresponding to the content identified as originating from a respective one of the plurality of acoustic zones, and performing automatic speech recognition on each of the plurality of second acoustic signals to determine whether the designated wake-up word was spoken.

Abstract: A method, computer program product, and computer system for identifying, by a computing device, a plurality of sources. One or more feature values of a plurality of features may be assigned to a first source of the plurality of sources. One or more feature values of the plurality of features may be assigned to a second source of the plurality of sources. A first score for the first source and a second score for the second source may be determined based upon, at least in part, the one or more feature values assigned to the first source and the second source. One of the first source and the second source may be selected for spatial processing based upon, at least in part, the first score for the first source and the second score for the second source.

Abstract: A system and method for generating a self-steering beamformer is provided. Embodiments may include receiving, at one or more microphones, a first audio signal and adapting one or more blocking filters based upon, at least in part, the first audio signal. Embodiments may also include generating, using the one or more blocking filters, one or more noise reference signals. Embodiments may further include providing the one or more noise reference signals to an adaptive interference canceller to reduce a beamformer output power level.

Abstract: Disclosed are systems and methods aware of ongoing conversations and configured to intelligently schedule a speech prompt to an intended addressee. A method for intelligently scheduling a speech prompt in a speech dialog system includes monitoring an acoustic environment to detect an intended addressee's availability for a speech prompt having a measure of urgency corresponding therewith. Based on the intended addressee's availability, the method predicts a time that is convenient to present the speech prompt to the intended addressee, and schedules the speech prompt based on the predicted time and the measure of urgency. A measure of rudeness can be estimated using a cost function that includes cost for presence of an utterance, cost for presence of a conversation, and cost for involvement of the intended addressee in the conversation. Scheduling the speech prompt can include trading off the measure of urgency and the measure of rudeness.

Abstract: A multi-mode voice controlled user interface is described. The user interface is adapted to conduct a speech dialog with one or more possible speakers and includes a broad listening mode which accepts speech inputs from the possible speakers without spatial filtering, and a selective listening mode which limits speech inputs to a specific speaker using spatial filtering. The user interface switches listening modes in response to one or more switching cues.

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: A method, computer program product, and computer system for identifying, by a computing device, a plurality of sources. One or more feature values of a plurality of features may be assigned to a first source of the plurality of sources. One or more feature values of the plurality of features may be assigned to a second source of the plurality of sources. A first score for the first source and a second score for the second source may be determined based upon, at least in part, the one or more feature values assigned to the first source and the second source. One of the first source and the second source may be selected for spatial processing based upon, at least in part, the first score for the first source and the second score for the second source.

Abstract: A method, computer program product, and computer system for receiving, by a computing device, a first signal emitted from one or more sources. A second signal may be received emitted from the one or more sources. A first confidence level that the wake-up-word is included in the first signal may be determined. A second confidence level that the wake-up-word is included in the second signal may be determined. It may be identified that the wake-up-word originated from a first source of the one or more sources based upon, at least in part, the first and second confidence levels. The first source may be enabled to participate in a dialog phase.

Abstract: A method, computer program product, and computer system for receiving, by a computing device, an input signal. A first power spectral density estimate may be generated for a linear reverberant component associated with the input signal. A second power spectral density estimate may be generated for a non-linear reverberant component associated with the input signal. A power spectral density estimate may be generated by combining the first power spectral density estimate for the linear reverberant component and the second power spectral density estimate for the non-linear reverberant component. One or more parameters for at least one of the linear reverberant component and the non-linear reverberant component may be updated. One or more undesired signal components in an output signal resulting from the input signal may be reduced via residual echo suppression based upon, at least in part, updating the one or more parameters.

Abstract: A system and method for speech enhancement of a portable electronic device. Embodiments may include receiving an audio signal at a portable electronic device having a first microphone and a second microphone. Embodiments may also include receiving an input from a proximity detector associated with the portable electronic device and controlling a processing component associated with at least one of the first microphone and the second microphone based upon, at least in part, the input from the proximity detector.

Abstract: A system and associated method for estimating one or more reverberation parameters includes an adaptive, parametric, linear prediction filter having one or more power spectral density (PSD) estimates of signals of one or more channels as inputs. The prediction filter estimates at least one parameter related to reverberation time, such as T60, and can further estimate an additional parameter, such as Direct-to-Reverberant Ratio (DRR). The prediction filter may be adapted during a period of reverberation by minimizing a cost function. Adaptation can include using a gradient descent approach, which can operate according to a step size provided by an adaptation controller configured to determine the period of reverberation. One or more microphones can provide the signals. The reverberation parameters estimated can be applied to a reverberation suppressor, with an estimator that does not require a training phase and without relying on assumptions of the user's position relative to the microphones.

Abstract: A method, computer program product, and computer system for receiving, by a computing device, a first signal emitted from one or more sources. A second signal may be received emitted from the one or more sources. A first confidence level that the wake-up-word is included in the first signal may be determined. A second confidence level that the wake-up-word is included in the second signal may be determined. It may be identified that the wake-up-word originated from a first source of the one or more sources based upon, at least in part, the first and second confidence levels. The first source may be enabled to participate in a dialogue phase.

Abstract: A method, computer program product, and computer system for addressing acoustic signal reverberation is provided. Embodiments may include receiving, at one or more microphones, a first audio signal and a reverberation audio signal. Embodiments may further include processing at least one of the first audio signal and the reverberation audio signal. Embodiments may also include limiting a model based reverberation equalizer using a temporal constraint for direct sound distortions, the model based reverberation equalizer configured to generate one or more outputs, based upon, at least in part, at least one of the first audio signal and the reverberation audio signal.

Abstract: A method, computer program product, and computer system for receiving, by a computing device, an input signal. A first power spectral density estimate may be generated for a linear reverberant component associated with the input signal. A second power spectral density estimate may be generated for a non-linear reverberant component associated with the input signal. A power spectral density estimate may be generated by combining the first power spectral density estimate for the linear reverberant component and the second power spectral density estimate for the non-linear reverberant component. One or more parameters for at least one of the linear reverberant component and the non-linear reverberant component may be updated. One or more undesired signal components in an output signal resulting from the input signal may be reduced via residual echo suppression based upon, at least in part, updating the one or more parameters.

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: A system and associated method for estimating one or more reverberation parameters includes an adaptive, parametric, linear prediction filter having one or more power spectral density (PSD) estimates of signals of one or more channels as inputs. The prediction filter estimates at least one parameter related to reverberation time, such as T60, and can further estimate an additional parameter, such as Direct-to-Reverberant Ratio (DRR). The prediction filter may be adapted during a period of reverberation by minimizing a cost function. Adaptation can include using a gradient descent approach, which can operate according to a step size provided by an adaptation controller configured to determine the period of reverberation. One or more microphones can provide the signals. The reverberation parameters estimated can be applied to a reverberation suppressor, with an estimator that does not require a training phase and without relying on assumptions of the user's position relative to the microphones.

Abstract: According to some aspects, a system for detecting a designated wake-up word is provided, the system comprising a plurality of microphones to detect acoustic information from a physical space having a plurality of acoustic zones, at least one processor configured to receive a first acoustic signal representing the acoustic information received by the plurality of microphones, process the first acoustic signal to identify content of the first acoustic signal originating from each of the plurality of acoustic zones, provide a plurality of second acoustic signals, each of the plurality of second acoustic signals substantially corresponding to the content identified as originating from a respective one of the plurality of acoustic zones, and performing automatic speech recognition on each of the plurality of second acoustic signals to determine whether the designated wake-up word was spoken.

Abstract: A method, computer program product, and computer system for receiving, by a computing device, a first signal emitted from one or more sources. A second signal may be received emitted from the one or more sources. A first confidence level that the wake-up-word is included in the first signal may be determined. A second confidence level that the wake-up-word is included in the second signal may be determined. It may be identified that the wake-up-word originated from a first source of the one or more sources based upon, at least in part, the first and second confidence levels. The first source may be enabled to participate in a dialog phase. The second source may be excluded from participating in the dialog phase.

Abstract: A system and method for acoustic echo cancellation is provided. Embodiments may include receiving, at one or more microphones, an audio reference signal from an audio speaker. Embodiments may also include filtering the audio reference signal using one or more adaptive audio filters. Embodiments may further include analyzing a level of signal energy of the audio reference signal with regard to time, frequency and audio channel to identify at least one maximum error contribution point. Embodiments may also include updating the one or more adaptive audio filters based upon, at least in part, the analyzed audio reference signal.