Abstract: Sound sources can be spatially rendered in a setting and shown through a display. In response to satisfaction of a threshold criterion that is satisfied based on relative distance between the sound sources and a position of a listener, the rendering of the sound sources can be adjusted to maintain spatial integrity of the sound sources. The adjustment can be performed to prevent one of the sound sources from arriving at the listener earlier than another of the sound sources.

Abstract: A method for self-calibrating a sound pickup process that uses a microphone array in a wearable device that also includes a loudspeaker, where the microphone array being in a physical arrangement with respect to the loudspeaker. The method obtains, for each of several microphones of the microphone array, one or more transfer functions that each represent a response of the microphone to sound from a position in an acoustic space. The method determines whether a physical arrangement of the microphone array with respect to the loudspeaker has changed and adjusts the transfer function, for at least one of the microphones of the several microphones, in response to determining that the current physical arrangement of the microphone array with respect to the loudspeaker has changed.

Abstract: In one implementation, a method of visualizing a combined audio pick-up pattern is performed at a first device in a physical environment, the first device including a display, one or more processors, and non-transitory memory. The method includes determining a first audio pick-up pattern of the first device. The method includes determining one or more second audio pick-up patterns of a respective one or more second devices. The method includes determining a combined audio pick-up pattern of the first device and the one or more second devices based on the first audio pick-up pattern and the one or more second audio pick-up patterns. The method includes displaying, on the display, a representation of the combined audio pick-up pattern. In one implementation, a method of determining an audio emission pattern is performed at a first device at a first location, the first device having a microphone, one or more processors, and non-transitory memory.

Abstract: Microphone signals of a primary headphone are processed and either a first transparency mode of operation is activated or a second transparency mode of operation. In another aspect, a processor enters different configurations in response to estimated ambient acoustic noise being lower or higher than a threshold, wherein in a first configuration a transparency audio signal is adapted via target voice and wearer voice processing (TVWVP) of a microphone signal to boost detected speech frequencies in the transparency audio signal, and in a second configuration the TVWVP is controlled to, as the estimated ambient acoustic noise increases, reduce boosting of, or not boost at all, the detected speech frequencies in the transparency audio signal. Other aspects are also described and claimed.

Abstract: A method performed by a first headset worn by a first user, the method includes the first headset performing noise cancellation on a microphone signal captured by a microphone of the first headset that is arranged to capture sounds within an ambient environment in which the first user is located. The first headset receives, from a second headset that is being worn by a second user who is in the ambient environment and over a wireless communication link, at least one sound characteristic generated by at least one sensor of the second headset and passing through select sounds from the microphone signal based on the received sound characteristic.

Abstract: A device with microphones can generate microphone signals during an audio recording. The device can store, in an electronic audio data file, the microphone signals, and metadata that includes impulse responses of the microphones. Other aspects are described and claimed.

Abstract: A multi-radius spherical microphone that includes an inner body defining an inner sphere having an inner radius from a center; a plurality of inner microphones coupled to the inner spherical body and defining an array of inner microphones; an outer body defining an dodecahedron, wherein the inner body and the outer body are concentric about the center; and a plurality of outer microphones coupled to the outer body at respective vertices of the dodecahedron and defining an array of outer microphones, wherein each of the plurality of outer microphones is positioned radially equidistant from the center.

Abstract: A method for self-calibrating a sound pickup process that uses a microphone array in a wearable device that also includes a loudspeaker, where the microphone array being in a physical arrangement with respect to the loudspeaker. The method obtains, for each of several microphones of the microphone array, one or more transfer functions that each represent a response of the microphone to sound from a position in an acoustic space. The method determines whether a physical arrangement of the microphone array with respect to the loudspeaker has changed and adjusts the transfer function, for at least one of the microphones of the several microphones, in response to determining that the current physical arrangement of the microphone array with respect to the loudspeaker has changed.

Abstract: Aspects of the subject technology provide for joint processing of signals from acoustic microphones and signals from vibration sensors that directly or remotely sense vibrations of the source of the sound itself. The vibration sensors may include remote vibration sensors such as a light-based microphone, which may be implemented as an optical microphone. Joint processing of the signals may include detecting a sound from the source in the signals from the acoustic microphone by selecting a portion of the signals from the acoustic microphone based on the signals from the vibration sensor.

Abstract: A method performed by a processor of a computer system including a headset that is to be worn on a head of a user. The method drives a speaker of the headset with an input audio signal to output sound into an environment. The method determines that the speaker is at least partially covered with a cupped hand. In response to determining that the speaker is at least partially covered with the cupped hand, applying a gain to the input audio signal to reduce an output sound level of the speaker.

Abstract: In a device or method for rendering a sound program for headphones, a location is received for placing the sound program with respect to first and second ear pieces. If the location is between the first ear piece and the second ear piece, the sound program is filtered to produce low-frequency and high-frequency portions. The high-frequency portion is panned according to the location to produce first and second high-frequency signals. The low-frequency portion and the first high-frequency signal are combined to produce a first headphone driver signal to drive the first ear piece. A second headphone driver signal is similarly produced. The sound program may be a stereo sound program. The device or method may also provide for a location that is between the first ear piece and a near-field boundary. Other aspects are also described.

Abstract: A method performed by a processor of an electronic device. The method presents a computer-generated reality (CGR) setting including a first user and several other users. The method obtains, from a microphone, an audio signal that contains speech of the first user. The method obtains, from a sensor, sensor data that represents a physical characteristic of the first user. The method determines, based on the sensor data, whether to initiate a private conversation between the first user and a second user of the other users, and in accordance with a determination to initiate the private conversation, initiates the private conversation by providing the audio signal to the second user.

Abstract: A plurality of microphone signals of a microphone array may be obtained. An environment change may be detected based on the microphone signals. In response, a reverberation time environment may be determined. The reverberation may be used to modify a playback audio signal.

Abstract: Disclosed are techniques for a multimedia device with audio and video capturing capability to identify an audio device based on acoustic playback signal if the audio device cannot be identified from captured video. The multimedia device may assemble a list of candidate audio devices that are a possible match for the observed audio device from a database of previously recognized audio devices and may transmit commands to the candidate audio devices to play acoustic identification signals. The acoustic identification signals may be audible sound or ultrasonic tone sequences with embedded identification information unique to each audio device. The multimedia device may record and analyze the acoustic identification signals received from any of the candidate audio devices to construct metrics to select the most likely candidate for the observed audio device. The metrics may include time of flight, direction of arrival, received amplitude, direct-to-reverberant ratio (DRR) of the acoustic identification signals.

Abstract: Audio processing with audio transparency can include receiving a user content audio signal and receiving a microphone signal. The microphone signal can contain sensed sound of a user environment. Strength of the sensed sound can be increased based on strength of the user content audio signal, to reduce a masking of the sensed sound during playback. The sensed sound and the user content audio signal can be combined in a composite output audio signal used to drive a speaker. Other aspects are also described and claimed.

Abstract: A method for routing audio content through an electronic device that is to be worn by a user. The method obtains a communication and determines whether the communication is private. In response to determining that the communication is private, the method drives a bone conduction transducer of the electronic device with an audio signal associated with the communication. In response to determining that the communication is not private, however, the method drives a speaker of the electronic device with the audio signal.

Abstract: Ray tracing is performed with sound sources and a listener position in a listening environment, to generate impulse responses associated with each of the sound sources. The impulse responses are combined to form a combined impulse response. One or more filters are determined, each corresponding to the sound sources, based on the combined impulse response and the impulse responses. Each filter serves as a correction factor that holds unique acoustic information for the sound source that the filter is associated with. The combined impulse response and the filters can be applied to one or more audio signals that contain the sound sources, resulting in audio having reverberation that is tailored to the various sound sources. Other aspects are described and claimed.

Abstract: One or more acoustic parameters of a current acoustic environment of a user may be determined based on sensor signals captured by one or more sensors of the device. One or more preset acoustic parameters may be determined based on the one or more acoustic parameters of the current acoustic environment of the user and an acoustic environment of an audio file comprising audio signals that is determined based on the audio signals of the audio file or metadata of the audio file. The audio signals may be spatially rendered by applying spatial filters that include the one or more preset acoustic parameters to the audio signals, resulting in binaural audio signals. The binaural audio signals may be used to drive speakers of a headset. Other aspects are described and claimed.

Abstract: A method for routing audio content through an electronic device that is to be worn by a user. The method obtains a communication and determines whether the communication is private. In response to determining that the communication is private, the method drives a bone conduction transducer of the electronic device with an audio signal associated with the communication. In response to determining that the communication is not private, however, the method drives a speaker of the electronic device with the audio signal.

Abstract: Disclosed are systems and methods for automatically transitioning between communication modes of wearable audio output devices based solely on acoustic analysis. The audio output devices may operate in one of three electroacoustic modes. In the transparency mode, an audio output device may pass through the speech signal of a nearby user. In the peer-to-peer mode, the audio output device may establish a direct low-latency radio frequency (RF) link to another audio output device. In the telephony mode, the audio output device may communicate with another audio output device using networked telephony. The disclosed methods and systems perform acoustic analysis of the near-field speech signal of a local wearer of the audio output device and the far-field speech signal of a remote talker to determine the best mode for the audio output device to use and to seamlessly transition between the modes as the acoustic environment between the wearers changes.