Techniques for adaptive noise cancellation for multiple audio endpoints in a shared space are described. According to one example, a method includes detecting, by a first audio endpoint, one or more audio endpoints co-located with the first audio endpoint at a first location. A selected audio endpoint of the one or more audio endpoints is identified as a target noise source. The method includes obtaining, from the selected audio endpoint, a loudspeaker reference signal associated with a loudspeaker of the selected audio endpoint and removing the loudspeaker reference signal from a microphone signal associated with a microphone of the first audio endpoint. The method also includes providing the microphone signal from the first audio endpoint to at least one of a voice user interface (VUI) or a second audio endpoint, wherein the second audio endpoint is located remotely from the first location.

An audio cancellation system includes a voice enabled computing system that is connected to an audio output device using a wired or wireless communication network. The voice enabled computing device can provide media content to a user and receive a voice command from the user. The connection between the voice enabled computing system and the audio output device introduces a time delay between the media content being generated at the voice enabled computing device and the media content being reproduced at the audio output device. The system operates to determine a calibration value adapted for the voice enabled computing system and the audio output device. The system uses the calibration value to filter the user's voice command from a recording of ambient sound including the media content, without requiring significant use of memory and computing resources.

One example includes a digital twin of a microphone array. The digital twin acts as a digital copy of a physical microphone array. The digital array allows the microphone array to be analyzed, simulated and optimized. Further, the microphone array can be optimized for performing sound quality operations such as noise suppression and speech intelligibility.

A collaborative distributed microphone array is configured to perform or be used in sound quality operations. A distributed microphone array can be operated to provide sound quality operations including sound suppression operations and speech intelligibility operations for multiple users in the same environment.

Provided a robot for assisting hearing of a user, while minimizing an influence on the surroundings. The robot includes a speaker, a microphone configured to recognize a voice, a processor configured to acquire a position of a user's face when a hearing aid command is acquired on the basis of the voice recognized through the microphone, and a driving unit configured to cause the speaker to be moved toward the position of the user's face, wherein the processor acquires a sound which is a target of hearing aid, generates an assistant sound by amplifying a predetermined frequency band of the sound or converting the predetermined frequency band of the sound into a different frequency band, and outputs the assistant sound through the speaker.

There is provided an automated microphone system (100) having a microphone (104). The system (100) comprises a microphone stand (102) having at least one movable arm (1022), at least one movable leg (1024) and a fixed base (1026) attached to the at least one movable leg (1024), the at least one movable arm (1022) being adapted to mount the microphone (104) at a first end and connected with the at least one movable leg (1024) at a second end, one or more sensors (106) disposed at one or more of the at least one movable leg (1024), the at least one movable arm (1022) and the fixed base (1026), one or more motors (108) connected with each of the at least one movable arm (1022), the at least one movable leg (1024) and the fixed base (1026) and a control module (110) connected with the one or more sensors (106) and the one or more motors (108).

An audio cancellation system includes a voice enabled computing system that is connected to an audio output device using a wired or wireless communication network. The voice enabled computing device can provide media content to a user and receive a voice command from the user. The connection between the voice enabled computing system and the audio output device introduces a time delay between the media content being generated at the voice enabled computing device and the media content being reproduced at the audio output device. The system operates to determine a calibration value adapted for the voice enabled computing system and the audio output device. The system uses the calibration value to filter the user's voice command from a recording of ambient sound including the media content, without requiring significant use of memory and computing resources.

Methods for rendering audio for playback by two or more speakers are disclosed. The audio includes one or more audio signals, each with an associated intended perceived spatial position. Relative activation of the speakers may be a cost function of a model of perceived spatial position of the audio signals when played back over the speakers, a measure of proximity of the intended perceived spatial position of the audio signals to positions of the speakers, and one or more additional dynamically configurable functions. The dynamically configurable functions may be based on at least one or more properties of the audio signals, one or more properties of the set of speakers and/or one or more external inputs.

Systems and methods are disclosed for networked audio automixing using array microphones and an aggregator unit that participate in making a common gating decision to determine which channels to gate on and off. Through the use of such a network of array microphones having the capability to generate submix audio signals and reduced bandwidth metrics, as well as AEC processing capability, array microphone lobe selection can be enhanced while maximizing signal-to-noise ratio, increasing intelligibility, and increasing user satisfaction.

Systems and methods are disclosed for operating a wireless audio network including a plurality of wireless microphone units (e.g., wireless delegate units) and a central access point having a mixer. The wireless microphone units may perform voice detection and level sensing, and make a preliminary gating decision. The central access point may make a final gating decision, determine the granting of wireless communications channels, and generate a final mixed audio output signal.