Abstract: A system controls the audio focus for a user in a virtual conference call. The system retrieves sound parameters associated with participants in the virtual conference call with the user from a user profile of the user. The sound parameters define volume adjustments to be applied to audio data received from the participants for generating an audio mix customized to the user. The system receives the audio data from client devices of the participants, and for each of the participants, adjusts the audio data of the participant using the associated sound parameter of the participant. The system adds the adjusted audio data of the participants to the audio mix for the user and provides the audio mix to a client device of the user.

Abstract: The disclosed computer-implemented method may include receiving, from a third party, a portion of data or computer-executable logic that is part of a specified model. Each model may include various portions of independently verifiable computer-executable logic. The method may further include receiving data at a processing engine. The processing engine may be configured to apply the specified model to the received data. The method may then execute the specified model at the processing engine to modify the received data and send the modified data to an application that is configured to process the modified data. Various other methods, systems, and computer-readable media are also disclosed.

Abstract: Determination of a set of acoustic parameters for a headset is presented herein. The set of acoustic parameters can be determined based on a virtual model of physical locations stored at a mapping server. The virtual model describes a plurality of spaces and acoustic properties of those spaces, wherein the location in the virtual model corresponds to a physical location of the headset. A location in the virtual model for the headset is determined based on information describing at least a portion of the local area received from the headset. The set of acoustic parameters associated with the physical location of the headset is determined based in part on the determined location in the virtual model and any acoustic parameters associated with the determined location. The headset presents audio content using the set of acoustic parameters received from the mapping server.

Abstract: Embodiments relate to an audio system for various audio applications. The audio system registers the locations of one or more sound sources and selects the target sound source based on a hidden Markov model. A health monitoring system that integrates an audio system may use information collected by sensors to monitor an amount of social interaction of a user and predict a risk of dementia and/or hearing loss based on a model. The audio system uses a current/voltage sensor to detect electrical drive signals for determining a level of audio leakage of the audio system. Additionally, the audio system may update a video stream with an audio background based on an artificial visual background in the video stream so that the updated video stream sounds as if it originated from the user being located in a physical representation related to the background.

Abstract: An audio system that includes an IMU or accelerometer in physical proximity to a microphone and a driver. The signal generated by the IMU or accelerometer may be used to filter the audio signal generated by the microphone with respect to mechanically coupling between the driver outputting sound and the microphone via a shared structure. In some cases, an accelerometer may be incorporated into a package of the microphone to provide analog-based filtering prior to converting the audio signal of the microphone to a digital format.

Abstract: A method comprising determining a set of position parameters for an inertial measurement unit (IMU) on a headset worn by a user. The set of position parameters includes at least a first yaw measurement and a first roll measurement. The set describes a pointing vector. The method further comprises calculating a drift correction component that describes a rate of correction. The drift correction component is based at least in part on the set of position parameters. The method further comprises applying the drift correction component to one or more subsequent yaw measurements for the IMU. The drift correction component forces an estimated nominal position vector to the pointing vector at the rate of correction.

Abstract: An electronic system is described that is responsive to non-verbal commands. The system may include a display component for presenting visual content to a user and an audio system, such as an in-ear or over-the-ear speaker system. The speaker system may be equipped with a bone conduction sensor or in-ear microphone that generates data associated with vibrations of a facial region of the user. The system may use the data to detect non-verbal command in the form of grunts, hums, coughs, tongue clicks, and the like. The system may perform various predetermined operations based on the detected vibrations.

Abstract: A system receives audio data in a frequency range of 20 Hz-20 kHz. The received audio data is encoded by the system into ultrasonic data in frequencies that are greater than 20 kHz, and transmitted into a local area that is proximal to the transmitting device, i.e., within the transmission range of the transmitting device. An ultrasonic communication device that is located in the transmission range of the transmitting device may receive the ultrasonic data. The received ultrasonic data is decoded by the ultrasonic communication system in the receiving device into audio data in a frequency range of 20 Hz-20 kHz, and subsequently presented to a user of the receiving ultrasonic communication device.

Abstract: The disclosed computer-implemented method may include applying, via a sound reproduction system, sound cancellation that reduces an amplitude of various sound signals. The method further includes identifying, among the sound signals, an external sound whose amplitude is to be reduced by the sound cancellation. The method then includes analyzing the identified external sound to determine whether the identified external sound is to be made audible to a user and, upon determining that the external sound is to be made audible to the user, the method includes modifying the sound cancellation so that the identified external sound is made audible to the user. Various other methods, systems, and computer-readable media are also disclosed.

Abstract: This disclosure describes embodiments of methods, non-transitory computer-readable media, and systems for detecting that a physical space includes a physical object corresponding to an analogous virtual object from an augmented reality experience and rendering (or otherwise modifying) the augment reality experience to integrate the physical object as part of the experience. In particular, the disclosed systems can determine that a physical object within a physical environment corresponds to an analogous virtual object of an augmented reality experience. Based on this correspondence, the disclosed systems can modify one or more of the virtual graphics, sound, or other features corresponding to the augmented reality experience to represent the virtual object using the physical object. For example, the disclosed systems can modify acoustic features of a sound for the augmented reality experience to simulate the sound originating from (or being affected by) the physical object.

Abstract: Determination of a set of acoustic parameters for a headset is presented herein. The set of acoustic parameters can be determined based on a virtual model of physical locations stored at a mapping server. The virtual model describes a plurality of spaces and acoustic properties of those spaces, wherein the location in the virtual model corresponds to a physical location of the headset. A location in the virtual model for the headset is determined based on information describing at least a portion of the local area received from the headset. The set of acoustic parameters associated with the physical location of the headset is determined based in part on the determined location in the virtual model and any acoustic parameters associated with the determined location. The headset presents audio content using the set of acoustic parameters received from the mapping server.

Abstract: An audio playback device detects via an acoustic sensor of an audio playback device, ambient noise surrounding the audio playback device. The audio playback device updates an equalization filter based on the detected ambient noise, wherein the equalization filter adjusts one or more acoustic parameters of content presented by the audio playback device. The audio playback device adjusts a leakage control parameter based on the detected ambient noise, wherein the leakage control parameter adjusts an amount of leakage of content presented by the audio playback device. The audio playback device applies the equalization filter and the adjusted leakage control parameter to audio content. The audio playback device provides, via one or more acoustic speakers, the audio content to a user.

Abstract: Determination of a set of acoustic parameters for a headset is presented herein. The set of acoustic parameters can be determined based on a virtual model of physical locations stored at a mapping server. The virtual model describes a plurality of spaces and acoustic properties of those spaces, wherein the location in the virtual model corresponds to a physical location of the headset. A location in the virtual model for the headset is determined based on information describing at least a portion of the local area received from the headset. The set of acoustic parameters associated with the physical location of the headset is determined based in part on the determined location in the virtual model and any acoustic parameters associated with the determined location. The headset presents audio content using the set of acoustic parameters received from the mapping server.

Abstract: An audio playback device detects via an acoustic sensor of an audio playback device, ambient noise surrounding the audio playback device. The audio playback device updates an equalization filter based on the detected ambient noise, wherein the equalization filter adjusts one or more acoustic parameters of content presented by the audio playback device. The audio playback device adjusts a leakage control parameter based on the detected ambient noise, wherein the leakage control parameter adjusts an amount of leakage of content presented by the audio playback device. The audio playback device applies the equalization filter and the adjusted leakage control parameter to audio content. The audio playback device provides, via one or more acoustic speakers, the audio content to a user.

Abstract: A headset generates an output audio signal to provide a virtual sound source for an object or virtual object by using a room impulse response generated by a simulation using a model of a room generated from image data. The headset may include processing circuitry that obtains the model of the room determined based on the image data. The image data includes depth image data from a depth camera assembly and color image data from a color camera. The model includes surfaces of the room and acoustic absorptions of the surfaces. The processing circuitry adjusts audio content presented by the headset based on a room impulse response determined based on one or more simulations of sound propagation between a target position of an object and a position of the headset within the room using the surfaces of the room and the acoustic absorptions of the surfaces.

Abstract: The disclosed computer-implemented method for performing natural language translation in AR may include accessing an audio input stream that includes words spoken by a speaking user in a first language. The method may next include performing active noise cancellation on the words in the audio input stream so that the spoken words are suppressed before reaching a listening user. Still further, the method may include processing the audio input stream to identify the words spoken by the speaking user, and translating the identified words spoken by the speaking user into a second, different language. The method may also include generating spoken words in the second, different language using the translated words, and replaying the generated spoken words in the second language to the listening user. Various other methods, systems, and computer-readable media are also disclosed.

Abstract: A system creates an audio profile. The audio profile may be stored in a database. For example, the audio profile may be securely stored in a database of a social network and associated with a user account. The audio profile may contain data describing the way in which the specific user hears and interprets sounds. Systems and applications which present sounds to the user may access the audio profile and modify the sounds presented to the user based on the data in the audio profile to enhance the audio experience for the user.

Abstract: A system reduces power consumption by optimizing a selection of acoustic sensors of a sensor array based on environmental parameters of a local area. The system includes the sensor array including the acoustic sensors configured to detect sound in a local area, and processing circuitry. The processing circuitry is configured to: determine an environmental parameter of the local area; determine a performance metric for the sensor array; determine a selection of a subset of acoustic sensors from the acoustic sensors of the sensor array that satisfies the performance metric based on the environmental parameter of the local area; and process audio data from the subset of the acoustic sensors of the sensor array.

Abstract: The disclosed computer-implemented method may include (1) establishing a communication channel to indirectly convey a conversation, (2) receiving, via the communication channel, a portion of the conversation, (3) presenting the portion of the conversation to a user, (4) receiving, via the communication channel, an additional portion of the conversation, (5) detecting an additional communication channel capable of conveying the conversation, (6) determining a human-perceivable difference between how the conversation has been conveyed via the communication channel and how the conversation will be conveyed via the additional communication channel, and (7) compensating for the human-perceivable difference when presenting the additional portion of the conversation to the user in order to smoothly transition the conversation from the communication channel to the additional communication channel. Various other methods, systems, and computer-readable media are also disclosed.

Abstract: A system creates an audio profile. The audio profile may be stored in a database. For example, the audio profile may be securely stored in a database of a social network and associated with a user account. The audio profile may contain data describing the way in which the specific user hears and interprets sounds. Systems and applications which present sounds to the user may access the audio profile and modify the sounds presented to the user based on the data in the audio profile to enhance the audio experience for the user.