Abstract: Techniques are described for own-voice suppression using separate sound filters. Sound from a local area is captured using an acoustic sensor array. The captured sound includes noise from the local area and the user's voice. A first set of statistical properties associated with the noise and a second set of statistical properties associated with the user's voice are determined. A first sound filter is generated based on a target transfer function and both sets of statistical properties. A second sound filter is generated based on the target transfer function and the second set of statistical properties, but not the first set of statistical properties. The first sound filter and the second sound filter are applied to different parts of an audio signal generated based on the captured sound, thereby forming audio content in which the user's voice is suppressed for presentation of the audio content to the user.

Abstract: A communication system provides audio content to one or more client devices capable of playing spatialized audio content. For example, the communication system receives audio content from a client device and transmits the audio content to other client devices to be played for users. The communication system dynamically modifies audio content transmitted to different client devices based on a payload including audio parameters (e.g., local area acoustic properties, an audiogram for a user, a head related transfer function for a user, etc.) received from a client device.

Abstract: A communication system provides audio content to one or more client devices capable of playing spatialized audio content. For example, the communication system receives audio content from a client device and transmits the audio content to other client devices to be played for users. The communication system dynamically modifies audio content transmitted to different client devices based on a payload including audio parameters (e.g., local area acoustic properties, an audiogram for a user, a head related transfer function for a user, etc.) received from a client device.

Abstract: An audio system presents enhanced audio content to a user of a headset. The audio system detects sounds from the local area, at least a portion of which originate from a human sound source. The audio system obtains a voice profile of an identifies human sound source that generates at least the portion of the detected sounds. Based in part on the voice profile, the audio system enhances the portion of the detected sounds that are generated by the human sound source to obtain enhanced audio. The audio system presents the enhanced audio to the user.

Abstract: A communication system provides audio content to one or more client devices capable of playing spatialized audio content. For example, the communication system receives audio content from a client device and transmits the audio content to other client devices to be played for users. The communication system dynamically modifies audio content transmitted to different client devices based on a payload including audio parameters (e.g., local area acoustic properties, an audiogram for a user, a head related transfer function for a user, etc.) received from a client device.

Abstract: An electronic device localizes an audio source by normalizing an amplitude of an audio signal over a time period. The electronic device receives, from one or more microphones of the electronic device, signal(s) representative of audio emitted by an audio source over a time period. The electronic device estimates amplitudes of the signal(s) at a first time within the time period and at a second time within the time period, where the second time is different from the first time. The electronic device normalizes the amplitudes associated with the first and second times to generate normalized amplitudes. The electronic device determines a combined amplitude representative of the audio emitted by the audio source by combining the normalized amplitudes. The electronic device determines, based at least in part on the combined amplitude and motion of the electronic device, an estimated position of the audio source relative to the electronic device.

Abstract: A system and method for storing data samples in discrete poses and recalling the stored data samples for updating a sound filter. The system determines that a microphone array at a first time period is in a first discrete pose of a plurality of discrete poses, wherein the plurality of discrete poses discretizes a pose space. The pose space includes at least an orientation component and may further include a translation component. The system retrieves one or more historical data samples associated with the first discrete pose, generated from sound captured by the microphone array before the first time period, and stored in a memory cache (e.g., for memorization). The system updates a sound filter for the first discrete pose using the retrieved one or more historical data samples. The system generates and presents audio content using the updated sound filter.

Abstract: The disclosed computer-implemented method may include receiving a signal for each channel of an audio transducer array on a wearable device. The method may also include calculating a beamformed signal for each beam direction of a set of beamforming filters for the wearable device. Additionally, the method may include classifying a first beamformed signal from the calculated beamformed signals into a first class of sound and a second beamformed signal from the calculated beamformed signals into a second class of sound. The method may also include adjusting, based on the classifying, a gain of the first beamformed signal relative to the second beamformed signal. Furthermore, the method may include converting the beamformed signals into spatialized binaural audio based on a position of a user. Finally, the method may include transmitting the spatialized binaural audio to a playback device. Various other methods, systems, and computer-readable media are also disclosed.

Abstract: A wearable device includes an audio system. In one embodiment, the audio system includes a sensor array that includes a plurality of acoustic sensors. When a user wears the wearable device, the audio system determines an acoustic transfer function for the user based upon detected sounds within a local area surrounding the sensor array. Because the acoustic transfer function is based upon the size, shape, and density of the user's body (e.g., the user's head), different acoustic transfer functions will be determined for different users. The determined acoustic transfer functions are compared with stored acoustic transfer functions of known users in order to authenticate the user of the wearable device.

Abstract: An eyewear device includes an audio system. In one embodiment, the audio system includes a microphone array that includes a plurality of acoustic sensors. Each acoustic sensor is configured to detect sounds within a local area surrounding the microphone array. For a plurality of the detected sounds, the audio system performs a direction of arrival (DoA) estimation. Based on parameters of the detected sound and/or the DoA estimation, the audio system may then generate or update one or more acoustic transfer functions unique to a user. The audio system may use the one or more acoustic transfer functions to generate audio content for the user.

Abstract: A headset generates an acoustic map that is a visual representation of sound pressure of sound within a local area. The acoustic map is displayed as a virtual object that is overlaid onto visual content being presented by the headset. Sound is adjusted in accordance with a command. The acoustic map is updated based on the adjusted sound, and the adjusted sound is presented and the updated acoustic map is presented.

Abstract: An audio system for a wearable device dynamically updates acoustic transfer functions. The audio system is configured to estimate a direction of arrival (DoA) of each sound source detected by a microphone array relative to a position of the wearable device within a local area. The audio system may track the movement of each sound source. The audio system may form a beam in the direction of each sound source. The audio system may identify and classify each sound source based on the sound source properties. Based on the DoA estimates, the movement tracking, and the beamforming, the audio system generates or updates the acoustic transfer functions for the sound sources.

Abstract: An audio system for near and far field signal enhancement that uses a contact transducer. The audio system includes a microphone array, the contact transducer, and a controller. The microphone array detects sounds from a local area. The sounds from the local area include a voice of a user of the audio system. The contact transducer may be in contact with the head of the user. The contact transducer detects tissue based vibrations that are generated by the user and pass through tissue of the user prior to being detected by the contact transducer. The controller identifies the voice of the user in the detected sounds using the detected tissue based vibrations and a model, and updates a sound filter based on the identified voice of the user. The audio content is modified using the updated sound filter, and is presented by at least one audio system.

Abstract: An electronic device localizes an audio source by normalizing an amplitude of an audio signal over a time period. The electronic device receives, from one or more microphones of the electronic device, signal(s) representative of audio emitted by an audio source over a time period. The electronic device estimates amplitudes of the signal(s) at a first time within the time period and at a second time within the time period, where the second time is different from the first time. The electronic device normalizes the amplitudes associated with the first and second times to generate normalized amplitudes. The electronic device determines a combined amplitude representative of the audio emitted by the audio source by combining the normalized amplitudes. The electronic device determines, based at least in part on the combined amplitude and motion of the electronic device, an estimated position of the audio source relative to the electronic device.

Abstract: An eyewear device includes an audio system. In one embodiment, the audio system includes a microphone array that includes a plurality of acoustic sensors. Each acoustic sensor is configured to detect sounds within a local area surrounding the microphone array. For a plurality of the detected sounds, the audio system performs a direction of arrival (DoA) estimation. Based on parameters of the detected sound and/or the DoA estimation, the audio system may then generate or update one or more acoustic transfer functions unique to a user. The audio system may use the one or more acoustic transfer functions to generate audio content for the user.

Abstract: An eyewear device includes an audio system. In one embodiment, the audio system includes a microphone array that includes a plurality of acoustic sensors. Each acoustic sensor is configured to detect sounds within a local area surrounding the microphone array. For a plurality of the detected sounds, the audio system performs a direction of arrival (DoA) estimation. Based on parameters of the detected sound and/or the DoA estimation, the audio system may then generate or update one or more acoustic transfer functions unique to a user. The audio system may use the one or more acoustic transfer functions to generate audio content for the user.

Abstract: An audio system for a wearable device dynamically updates acoustic transfer functions. The audio system is configured to estimate a direction of arrival (DoA) of each sound source detected by a microphone array relative to a position of the wearable device within a local area. The audio system may track the movement of each sound source. The audio system may form a beam in the direction of each sound source. The audio system may identify and classify each sound source based on the sound source properties. Based on the DoA estimates, the movement tracking, and the beamforming, the audio system generates or updates the acoustic transfer functions for the sound sources.

Abstract: The disclosed computer-implemented method may include receiving a signal for each channel of an audio transducer array on a wearable device. The method may also include calculating a beamformed signal for each beam direction of a set of beamforming filters for the wearable device. Additionally, the method may include classifying a first beamformed signal from the calculated beamformed signals into a first class of sound and a second beamformed signal from the calculated beamformed signals into a second class of sound. The method may also include adjusting, based on the classifying, a gain of the first beamformed signal relative to the second beamformed signal. Furthermore, the method may include converting the beamformed signals into spatialized binaural audio based on a position of a user. Finally, the method may include transmitting the spatialized binaural audio to a playback device. Various other methods, systems, and computer-readable media are also disclosed.

Abstract: An audio system for a wearable device dynamically updates acoustic transfer functions. The audio system is configured to estimate a direction of arrival (DoA) of each sound source detected by a microphone array relative to a position of the wearable device within a local area. The audio system may track the movement of each sound source. The audio system may form a beam in the direction of each sound source. The audio system may identify and classify each sound source based on the sound source properties. Based on the DoA estimates, the movement tracking, and the beamforming, the audio system generates or updates the acoustic transfer functions for the sound sources.

Abstract: One embodiment of the present application sets forth a computer-implemented method that includes receiving, from a first microphone, a first input acoustic signal, generating a first audio spectrum from at least the first input acoustic signal, wherein the first audio spectrum includes a set of time-frequency bins, and selecting a first time-frequency bin from the set based on a first local space-domain distance (LSDD) computed for the first time-frequency bin.