A method for audio beam steering, tracking, and audio effects for an immersive reality application is provided. The method includes receiving, from an immersive reality application, a first audio waveform from a first acoustic source to provide to a user of a headset, identifying a perceived direction for the first acoustic source relative to the headset based on a location of the first acoustic source, and providing, to a first speaker in a client device, an audio signal including the first audio waveform, wherein the audio signal includes a time delay and an amplitude of the first audio waveform based on the perceived direction. A non-transitory, computer-readable medium storing instructions which, when executed by a processor, cause a system to perform the above method, and the system, are also provided.

A method to assess user condition for wearable devices using electromagnetic sensors is provided. The method includes receiving a signal from an electromagnetic sensor, the signal being indicative of a health condition of a user of a wearable device, selecting a salient attribute from the signal, and determining, based on the salient attribute, the health condition of the user of the wearable device. A non-transitory, computer-readable medium storing instructions which, when executed by a processor, cause a system to perform the above method, and the system, are also provided.

A method to combine contact and acoustic microphones in a headset for voice wake and voice processing in immersive reality applications is provided. The method includes receiving, from a contact microphone, a first acoustic signal, determining a fidelity and a quality of the first acoustic signal, receiving, from an acoustic microphone, a second acoustic signal, and when the fidelity and quality of the first acoustic signal exceeds a pre-selected threshold, combining the first acoustic signal and the second acoustic signal to provide an enhanced acoustic signal to a smart glass user. A non-transitory, computer-readable medium storing instructions to cause a headset to perform the above method, and the headset, are also provided.

A system for sound localization can include a first electronic device having a microphone to detect a sound, and a second electronic device. A processor can be in communication with the first electronic device and the second electronic device. The processor can receive a first signal from the first electronic device corresponding to the detected sound, determine a location of origin of the detected sound based at least in part on the first signal, and provide a second signal to the second electronic device based at least in part on the location of origin.

Systems and methods for dynamic voice accentuation and reinforcement are presented herein. One embodiment comprises one or more audio input sources; one or more audio output sources; one or more band pass filters; and a processing control unit that includes an audio processing unit, and which executes a method: differentiating between audio input sources as vocal sound audio input sources and ambient noise audio input sources; increasing the gain of the vocal sound audio input sources; inverting a polarity of an ambient noise signal received by each of the ambient noise audio input sources; and adding the inverted polarity to either an output signal of at least one of the one or more audio output sources, or to an input signal of at least one of the vocal sound audio input sources, to reduce ambient noise.

Techniques and systems are described for generating and using a sound localization model. A described technique includes obtaining for a building a sound sensor map indicating locations of first and second sound sensor devices in respective first and second rooms of the building; causing an autonomous device to navigate to the first room and to emit, during a time window, sound patterns at one or more frequencies within the first room; receiving sound data including first and second sound data respectively from the first and second sound sensor devices that are observed during the time window; and generating and storing a sound localization model based on the sound sensor map, autonomous device location information, and the received sound data, the model being configured to compensate for how sounds travels among rooms in at least a portion of the building such that an origin room of a sound source is identifiable.

There is provided a smart audio system including multiple audio devices and a central server. The central server confirms a model of every audio device and a position thereof in an operation area in a scan mode. The central server confirms a user position or a user state to accordingly control output power of a speaker of each of the multiple audio devices in an operation mode.

A method, computer program product, and computing system for proactive encounter scanning is executed on a computing device and includes obtaining encounter information of a patient encounter. The encounter information is proactively processed to determine if the encounter information is indicative of one or more medical conditions and to generate one or more result set. The one or more result sets are provided to the user.

A method for estimating an acoustic influence of walls of a room, comprising emitting a known excitation sound signal, receiving a set of measurement signals, each measurement signal being received by one microphone in a microphone array and each measurement signal including a set of echoes caused by reflections by the walls, solving a linear system of equations to identify locations of image source and estimating the acoustic influence based these image sources. The signal model includes a convolution of: the excitation signal, a multichannel filter (M) representing the relative delays of the microphones in the microphone array, the relative delays determined based on a known geometry of the microphone array, and a directivity model ν(n, p) of the driver(s) in the form of an anechoic far-field impulse response as a function of transmit angle.

The present teachings relate to an electronic device comprising: a first module for generating an audio signal; a second module for generating an ultrasonic signal; a mixer for generating a combined signal; a transmitter for outputting an acoustic signal dependent upon the combined signal; and, a processing means for controlling the ultrasonic signal; wherein, in response to receiving a first instruction signal for initiating the ultrasonic signal, the processing means is configured to increase the amount of the ultrasonic signal in the combined signal from an essentially zero value to a predetermined value over a predetermined enable time-period. The present teachings also relate to an electronic device configured to decrease the amount of the ultrasonic signal in the combined signal from an essentially zero value to a predetermined value over a predetermined disable time-period, and to an electronic device configured to remove the audio signal from the combined signal whilst preventing pop-noise, and to an electronic device capable of replacing the ultrasonic signal whilst minimizing the processing time. The present teachings further relate to a method for reducing the occurrence of pop noise in an acoustic signal associated with: initiating the ultrasonic signal in the combined signal, terminating the ultrasonic signal in the combined signal, terminating the audio signal in the combined signal, and replacing the ultrasonic signal in the combined signal. The present teachings also relate to a computer software product for implementing any of the method steps disclosed herein, and to a computer storage medium storing the computer software herein disclosed.