Abstract: An audio processing circuit includes an audio front end (AFE) configured to generate audio data in response to audio signals received from one or more microphones. One or more special purpose engines (SPEs) are configured to: generate prediction data via a usage model; generate bias adjustment data based on the prediction data; adjust a threshold of an audio event recognition engine based on the prediction data; and determine, via the audio event recognition engine and based on audio data, when an audio event is detected.

Abstract: Systems, devices, and methods are described for providing loudspeaker protection and distortion suppression. An upstream loudspeaker model estimation component receives sensed electrical characteristics of a loudspeaker and generates an impedance model from which an excursion model of the loudspeaker and a gain change parameter may be generated. The impedance components are fitted to features of an estimated impedance, based on the sensed characteristics, to generate the estimated impedance model that is converted to an excursion model of the loudspeaker. A downstream audio signal processing component, based on the excursion model, or parameters thereof, limits a predicted excursion of the loudspeaker utilizing excursion-constraining processing circuitry that includes a non-linear constraint filter. Processed audio signals associated with the limited excursion are subject to distortion suppression prior to releasing the output audio signals for playback on the loudspeaker.

Abstract: Systems, devices, and methods are described for providing loudspeaker protection. An upstream loudspeaker model estimation component receives sensed electrical characteristics of a loudspeaker and generates an impedance model from which an excursion model, and associated parameters, of the loudspeaker as well as a gain change parameter may be generated. The impedance components are fitted to features of an estimated impedance, based on the voltage and current sense data, to generate the estimated impedance model that is converted to an excursion model of the loudspeaker. A downstream audio signal processing component, based on the excursion model, or parameters thereof, limits a predicted excursion of the loudspeaker utilizing excursion-constraining processing circuitry that includes a non-linear constraint filter. Processed audio signals associated with the limited excursion are subject to distortion suppression prior to releasing the output audio signals for playback on the loudspeaker.

Abstract: Systems, devices, and methods are described for providing loudspeaker protection. An upstream loudspeaker model estimation component receives sensed electrical characteristics of a loudspeaker, estimates the impedance of the loudspeaker, fits an impedance model to the estimated impedance, and generates an excursion model of the loudspeaker and a gain change parameter from the estimated parameters of the impedance model. A downstream audio signal processing component utilizes the gain change parameter and the excursion model, or parameters thereof, to constrain the temperature of a voice coil of the loudspeaker and to limit a predicted excursion of the loudspeaker. The downstream audio signal processing component also utilizes processed signals associated with the constrained temperature and the limited excursion to suppress unwanted distortion for an output audio signal to be played back by the loudspeaker.

Abstract: Systems, devices, and methods are described for providing loudspeaker protection. An upstream loudspeaker model estimation component receives sensed electrical characteristics of a loudspeaker and generates an impedance model from which an excursion model, and associated parameters, of the loudspeaker as well as a gain change parameter may be generated. The impedance components are fitted to features of an estimated impedance, based on the voltage and current sense data, to generate the estimated impedance model of the loudspeaker by combining the fitted impedance components. The resulting estimated impedance model is converted to an excursion model of the loudspeaker. A downstream audio signal processing component utilizes the excursion model, or parameters thereof, to limit a predicted excursion of the loudspeaker. Processed audio signals associated with the limited excursion are subject to distortion suppression prior to releasing the output audio signals for playback on the loudspeaker.

Abstract: Systems, devices, and methods are described for providing loudspeaker protection and distortion suppression. An upstream loudspeaker model estimation component receives sensed electrical characteristics of a loudspeaker and generates an impedance model from which an excursion model of the loudspeaker and a gain change parameter may be generated. The impedance components are fitted to features of an estimated impedance, based on the sensed characteristics, to generate the estimated impedance model that is converted to an excursion model of the loudspeaker. A downstream audio signal processing component, based on the excursion model, or parameters thereof, limits a predicted excursion of the loudspeaker utilizing excursion-constraining processing circuitry that includes a non-linear constraint filter. Processed audio signals associated with the limited excursion are subject to distortion suppression prior to releasing the output audio signals for playback on the loudspeaker.

Abstract: Systems, devices, and methods are described for providing loudspeaker protection. An upstream loudspeaker model estimation component receives sensed electrical characteristics of a loudspeaker and generates an impedance model from which an excursion model, and associated parameters, of the loudspeaker as well as a gain change parameter may be generated. The impedance components are fitted to features of an estimated impedance, based on the voltage and current sense data, to generate the estimated impedance model of the loudspeaker by combining the fitted impedance components. The resulting estimated impedance model is converted to an excursion model of the loudspeaker. A downstream audio signal processing component utilizes the excursion model, or parameters thereof, to limit a predicted excursion of the loudspeaker. Processed audio signals associated with the limited excursion are subject to distortion suppression prior to releasing the output audio signals for playback on the loudspeaker.

Abstract: Systems, devices, and methods are described for providing loudspeaker protection. An upstream loudspeaker model estimation component receives sensed electrical characteristics of a loudspeaker and generates an impedance model from which an excursion model, and associated parameters, of the loudspeaker as well as a gain change parameter may be generated. The impedance components are fitted to features of an estimated impedance, based on the voltage and current sense data, to generate the estimated impedance model that is converted to an excursion model of the loudspeaker. A downstream audio signal processing component, based on the excursion model, or parameters thereof, limits a predicted excursion of the loudspeaker utilizing excursion-constraining processing circuitry that includes a non-linear constraint filter. Processed audio signals associated with the limited excursion are subject to distortion suppression prior to releasing the output audio signals for playback on the loudspeaker.

Abstract: Systems, devices, and methods are described for providing loudspeaker protection. An upstream loudspeaker model estimation component receives sensed electrical characteristics of a loudspeaker, estimates the impedance of the loudspeaker, fits an impedance model to the estimated impedance, and generates an excursion model of the loudspeaker and a gain change parameter from the estimated parameters of the impedance model. A downstream audio signal processing component utilizes the gain change parameter and the excursion model, or parameters thereof, to constrain the temperature of a voice coil of the loudspeaker and to limit a predicted excursion of the loudspeaker. The downstream audio signal processing component also utilizes processed signals associated with the constrained temperature and the limited excursion to suppress unwanted distortion for an output audio signal to be played back by the loudspeaker.

Abstract: Techniques described herein are directed to performing back-end single-channel suppression of one or more types of interfering sources (e.g., additive noise) in an uplink path of a communication device. The back-end single-channel suppression techniques may suppress types(s) of additive noise using one or more suppression branches (e.g., a non-spatial (or stationary noise) branch, a spatial (or non-stationary noise) branch, a residual echo suppression branch, etc.). The non-spatial branch may be configured to suppress stationary noise from the single-channel audio signal, the spatial branch may be configured to suppress non-stationary noise from the single-channel audio signal and the residual echo suppression branch may be configured to suppress residual echo from the signal-channel audio signal. The spatial branch may be disabled based on an operational mode (e.g.

Abstract: Methods, systems, and apparatuses are described for improved multi-microphone source tracking and noise suppression. In multi-microphone devices and systems, frequency domain acoustic echo cancellation is performed on each microphone input, and microphone levels and sensitivity are normalized. Methods, systems, and apparatuses are also described for improved acoustic scene analysis and source tracking using steered null error transforms, on-line adaptive acoustic scene modeling, and speaker-dependent information. Switched super-directive beamforming reinforces desired audio sources and closed-form blocking matrices suppress desired audio sources based on spatial information derived from microphone pairings. Underlying statistics are tracked and used to updated filters and models. Automatic detection of single-user and multi-user scenarios, and single-channel suppression using spatial information, non-spatial information, and residual echo are also described.

Abstract: A speech intelligibility enhancement (SIE) system and method is described that improves the intelligibility of a speech signal to be played back by an audio device when the audio device is located in an environment with loud acoustic background noise. In an embodiment, the audio device comprises a near-end telephony terminal and the speech signal comprises a speech signal received over a communication network from a far-end telephony terminal for playback at the near-end telephony terminal.

Abstract: A system and method is described that improves the intelligibility of a far-end telephone speech signal to a user of a telephony device in the presence of near-end background noise. As described herein, the system and method improves the intelligibility of the far-end telephone speech signal in a manner that does not require user input and that minimizes the distortion of the far-end telephone speech signal. The system is integrated with an acoustic echo canceller and shares information therewith.

Abstract: Methods, systems, and apparatuses are described for improved multi-microphone source tracking and noise suppression. In multi-microphone devices and systems, frequency domain acoustic echo cancellation is performed on each microphone input, and microphone levels and sensitivity are normalized. Methods, systems, and apparatuses are also described for improved acoustic scene analysis and source tracking using steered null error transforms, on-line adaptive acoustic scene modeling, and speaker-dependent information. Switched super-directive beamforming reinforces desired audio sources and closed-form blocking matrices suppress desired audio sources based on spatial information derived from microphone pairings. Underlying statistics are tracked and used to updated filters and models. Automatic detection of single-user and multi-user scenarios, and single-channel suppression using spatial information, non-spatial information, and residual echo are also described.

Abstract: A speech intelligibility enhancement (SIE) system and method is described that improves the intelligibility of a speech signal to be played back by an audio device when the audio device is located in an environment with loud acoustic background noise. In an embodiment, the audio device comprises a near-end telephony terminal and the speech signal comprises a speech signal received over a communication network from a far-end telephony terminal for playback at the near-end telephony terminal.

Abstract: Unlike sound based pressure waves that go everywhere, air turbulence caused by wind is usually a fairly local event. Therefore, in a system that utilizes two or more spatially separated microphones to pick up sound signals (e.g., speech), wind noise picked up by one of the microphones often will not be picked up (or at least not to the same extent) by the other microphone(s). Embodiments of methods and apparatuses that utilize this tact and others to effectively detect and suppress wind noise using multiple microphones that are spatially separated are described.

Abstract: Methods, systems, and apparatuses are described for performing speaker-identification-assisted speech processing. In accordance with certain embodiments, a communication device includes speaker identification (SID) logic that is configured to identify a user of the communication device and/or the identity of a far-end speaker participating in a voice call with a user of the communication device. Knowledge of the identity of the user and/or far-end speaker is then used to improve the performance of one or more speech processing algorithms implemented on the communication device.

Abstract: Methods, systems, and apparatuses are described for performing speaker-identification-assisted speech processing in an uplink path of a communication device. In accordance with certain embodiments, a communication device includes speaker identification (SID) logic that is configured to identify the identity of a near-end speaker. Knowledge of the identity of the near-end speaker is then used to improve the performance of one or more uplink speech processing algorithms implemented on the communication device.

Abstract: A technique for suppressing non-stationary noise, such as wind noise, in an audio signal is described. In accordance with the technique, a series of frames of the audio signal is analyzed to detect whether the audio signal comprises non-stationary noise. If it is detected that the audio signal comprises non-stationary noise, a number of steps are performed. In accordance with these steps, a determination is made as to whether a frame of the audio signal comprises non-stationary noise or speech and non-stationary noise. If it is determined that the frame comprises non-stationary noise, a first filter is applied to the frame and if it is determined that the frame comprises speech and non-stationary noise, a second filter is applied to the frame.

Abstract: A speech intelligibility enhancement (SIE) system and method is described that improves the intelligibility of a speech signal to be played back by an audio device when the audio device is located in an environment with loud acoustic background noise. In an embodiment, the audio device comprises a near-end telephony terminal and the speech signal comprises a speech signal received over a communication network from a far-end telephony terminal for playback at the near-end telephony terminal.