Abstract: An audio system is described that includes one or more speaker arrays that emit sound corresponding to one or more pieces of sound program content into associated zones within a listening area. Using parameters of the audio system (e.g., locations of the speaker arrays and the audio sources), the zones, the users, the pieces of sound program content, and the listening area, one or more beam pattern attributes may be generated. The beam pattern attributes define a set of beams that are used to generate audio beams for channels of sound program content to be played in each zone. The beam pattern attributes may be updated as changes are detected within the listening environment. By adapting to these changing conditions, the audio system is capable of reproducing sound that accurately represents each piece of sound program content in various zones.

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: A binaural room impulse response (BRIR) can be generated based on a position of a listener's head, and a plurality of head related impulse responses (HRIRs). Each of the plurality of HRIRs are selected for a respective one of a plurality of acoustic reflections which, when taken together, approximate reverberation of a room. Each of the acoustic reflections have a direction and a delay. The BRIR filter is applied to source audio to generate binaural audio output.

Abstract: Methods and apparatus for predictive head-tracked binaural audio rendering in which a rendering device renders multiple audio streams for different possible head locations based on head tracking data received from a headset, for example audio streams for the last known location and one or more predicted or possible locations, and transmits the multiple audio streams to the headset. The headset then selects and plays one of the audio streams that is closest to the actual head location based on current head tracking data. If none of the audio streams closely match the actual head location, two closest audio streams may be mixed. Transmitting multiple audio streams to the headset and selecting or mixing an audio stream on the headset may mitigate or eliminate perceived head tracking latency.

Abstract: An audio system and a method of using the audio system to determine an interaural head-related transfer function (HRTF) parameter, are described. The audio system can generate binaural recordings using microphones that are worn by a user in everyday scenarios. The audio system can measure interaural parameter values of selected segments of the recordings, and the measurements can be accumulated over time. The interaural HRTF parameter can be estimated based on the measurements. The interaural HRTF parameter can be used to adapt a generic HRTF to generate an individualized HRTF for the user. Other aspects are also described and claimed.

Abstract: A multi-way speaker array is disclosed that includes rings of transducers of different types. The rings of transducers may encircle the cabinet of the speaker array such that the speaker array is rotationally symmetric. The distance between rings of transducers may be based on a logarithmic scale. By separating rings of transducers using logarithmic spacing, denser transducer spacing at short wavelengths is achieved while limiting the number of transducers needed for longer wavelengths by spacing them in larger and larger logarithmic increments. Transducers with overlapping frequency ranges may be used in the speaker array to avoid initial dips or shortfalls in directivity for corresponding beam patterns.

Abstract: A system for enhancing audio including a plurality of sensors, an output device, and a processor in communication with the plurality of sensors and the output device. The processor is configured to process data captured by the plurality of sensors, and based on that, modify an output of the output device. The processor also is configured to determine whether there are a plurality of users associated with a video conferencing session, determine which user of the plurality of users is speaking, and enhance the audio or video output of the speaking user on the output device.

Abstract: An audio system and a method of using the audio system to generate a head-related transfer function (HRTF) map, are described. The audio system can determine one or more HRTF and corresponding HRTF locations along an azimuthal path of an azimuth extending around a head of a user. The HRTFs or HRTF locations can be measured, and other HRTFs or HRTF locations can be interpolated or extrapolated. The HRTF map can include the HRTFs assigned to HRTF locations along the azimuth. HRTFs from the HRTF map are used to render spatial audio to the user. Other aspects are also described and claimed.

Abstract: The present disclosure generally relates techniques for audio-assisted enrollment of biometric features. In some embodiments, methods and devices for assisting users with enrollment of biometric features, using spatial audio cues, are described.

Abstract: The present disclosure generally relates to user interfaces for managing spatial audio. Some exemplary techniques include user interfaces for transitioning between visual elements. Some exemplary techniques include user interfaces for previewing audio. Some exemplary techniques include user interfaces for discovering music. Some exemplary techniques include user interfaces for managing headphone transparency. Some exemplary techniques include user interfaces for manipulating multiple audio streams of an audio source.

Abstract: A directivity adjustment device that maintains a constant direct-to-reverberant ratio based on the detected location of a listener in relation to the speaker array is described. The directivity adjustment device may include a distance estimator, a directivity compensator, and an array processor. The distance estimator detects the distance between the speaker array and the listener. Based on this detected distance, the directivity compensator calculates a directivity index form a beam produced by the speaker array that maintains a predefined direct-to-reverberant sound energy ratio. The array processor receives the calculated directivity index and processes each channel of a piece of sound program content to produce a set of audio signals that drive one or more of the transducers in the speaker array to generate a beam pattern with the calculated directivity index.

Abstract: A system for enhancing audio including a plurality of sensors, an output device, and a processor in communication with the plurality of sensors and the output device. The processor is configured to process data captured by the plurality of sensors, and based on that, modify an output of the output device. The processor also is configured to determine whether there are a plurality of users associated with a video conferencing session, determine which user of the plurality of users is speaking, and enhance the audio or video output of the speaking user on the output device.

Abstract: A method for producing a diffuse field that is non-localizable and without timbre inaccuracies. A processor receives an audio bitstream containing an at least one surround channel. The surround channel is rendered as an at least one virtualized line array source. Timbre correction is applied to the virtualized line array source. Other aspects are also described and claimed.

Abstract: A directivity adjustment device that maintains a constant direct-to-reverberant ratio based on the detected location of a listener in relation to the speaker array is described. The directivity adjustment device may include a distance estimator, a directivity compensator, and an array processor. The distance estimator detects the distance between the speaker array and the listener. Based on this detected distance, the directivity compensator calculates a directivity index form a beam produced by the speaker array that maintains a predefined direct-to-reverberant sound energy ratio. The array processor receives the calculated directivity index and processes each channel of a piece of sound program content to produce a set of audio signals that drive one or more of the transducers in the speaker array to generate a beam pattern with the calculated directivity index.

Abstract: An audio system having a depth capturing device and a microphone array to respectively detect a point cloud and a local sound field, is described. The point cloud includes points corresponding to objects in a source space, and the local sound field includes sounds received in respective directions from the source space. The audio system includes one or more processors to generate a global sound field based on distances to the points and directions of the sounds. The global sound field includes virtual sound sources emitting respective sounds at respective points. A speaker can render the virtual sound sources to a user in a virtual space that corresponds to the source space.

Abstract: The present disclosure generally relates to user interfaces for managing spatial audio. Some exemplary techniques include user interfaces for transitioning between visual elements. Some exemplary techniques include user interfaces for previewing audio. Some exemplary techniques include user interfaces for discovering music. Some exemplary techniques include user interfaces for managing headphone transparency. Some exemplary techniques include user interfaces for manipulating multiple audio streams of an audio source.

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: 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 system for enhancing audio including a plurality of sensors, an output device, and a processor in communication with the plurality of sensors and the output device. The processor is configured to process data captured by the plurality of sensors, and based on that, modify an output of the output device. The processor also is configured to determine whether there are a plurality of users associated with a video conferencing session, determine which user of the plurality of users is speaking, and enhance the audio or video output of the speaking user on the output device.

Abstract: Image analysis of a video signal is performed to produce first metadata, and audio analysis of a multi-channel sound track associated with the video signal is performed to produce second metadata. A number of time segments of the sound track are processed, wherein each time segment is processed by either (i) spatial filtering of the audio signals or (ii) spatial rendering of the audio signals, not both, wherein for each time segment a decision was made to select between the spatial filtering or the spatial rendering, in accordance with the first and second metadata. A mix of the processed sound track and the video signal is generated. Other embodiments are also described and claimed.