A method, and system, of digital room correction for a device, such as a smart speaker, including a loudspeaker. The method comprises capturing audio from an environment local to the device, for example from one or more microphones of a smart speaker. The captured audio is then processed to recognize one or more categories of sound. A digital room correction procedure may then be controlled dependent upon recognition and/or analysis of at least one of the categories of sound.

A venue includes an exterior structure, a theater area inside the exterior structure, an audience seating area positioned in the theater area and facing the stage, and a mezzanine area positioned inside the exterior structure, outside of the theater area, and at least partially beneath the audience seating area. The theater area includes an interior surface that provides or supports a display. The display is an immersive display that extends along the interior surface such that the display is positioned in front of, to the sides, above, and behind at least a portion of the audience seating area.

The amount of far-field noise transmitted by a primary communication device in an open-plan office environment is reduced by defining an acoustic perimeter of reference microphones around the primary device. Reference microphones generate a reference audio input including far-field noise in the proximity of the primary device. The primary device generates a main audio input including the voice of the primary speaker as well as background noise. Reference audio input is compared to main audio input to identify the background noise portion of the main audio signal. A noise reduction algorithm suppresses the identified background noise in the main audio signal. The one or more reference microphones defining the acoustic perimeter may be included in separate microphone devices placed in proximity to the main desktop phone, microphones within other nearby desktop telephone devices, or a combination of both types of devices.

Systems and methods for optimizing voice detection via a network microphone device (NMD) based on a selected voice-assistant service (VAS) are disclosed herein. In one example, the NMD detects sound via individual microphones and selects a first VAS to communicate with the NMD. The NMD produces a first sound-data stream based on the detected sound using a spatial processor in a first configuration. Once the NMD determines that a second VAS is to be selected over the first VAS, the spatial processor assumes a second configuration for producing a second sound-data stream based on the detected sound. The second sound-data stream is then transmitted to one or more remote computing devices associated with the second VAS.

The systems and methods described relate to the concept that smart devices can be used to: sense various types of phenomena like sound, blue light exposure, RF and microwave radiation, and, in real-time, analyze, report and/or control outputs (e.g., displays or speakers). The systems are configurable and use standard computing devices, such as wearable electronics (e.g., smart glasses), tablet computers, and mobile phones to measure various frequency bands across multiple points, allowing a single user to visualize and/or adjust environmental conditions.

Examples described herein involve configuring a playback device based on distortion, such as that caused by a barrier. One implementation may involve causing the playback device to play audio content according to an existing playback configuration, determining an existing frequency response of the playback device in a given system, and determining whether a difference between the existing frequency response of the playback device in the given system and a predetermined frequency response for the playback device is greater than a predetermined distortion threshold. If it is determined that the difference between the existing frequency response of the playback device and the predetermined frequency response for the playback device is greater than the predetermined distortion threshold, then the existing playback configuration of the playback device is changed to an updated playback configuration of the playback device and the playback device plays audio content according to the updated playback configuration.

A method of active noise equalization in a system comprising a plurality of nodes is disclosed, wherein each node comprises at least one acoustic sensor and at least one acoustic actuator, and each node has an associated target spectral noise profile and an associated set of adaptive filter coefficients. The method comprises the steps of: i) at each node, receiving a reference acoustic signal; ii) at each node, generating an output acoustic signal based on the received reference acoustic signal and the associated set of adaptive filter coefficients for the node; iii) at each node, receiving a measured acoustic signal for the node and computing a pseudo-error signal for the node in dependence upon the received measured acoustic signal and the target spectral noise profile associated with the node; and iv) updating at least one adaptive filter coefficient in the set of adaptive filter coefficients for at least a first node comprised in the plurality of nodes, in dependence upon the pseudo-error signal associated with the first node, the reference acoustic signal, and at least one parameter received from at least one other node in the plurality of nodes which is dependent on the pseudo-error signal associated with the at least one other node.

Reducing audio-based distractions in an information handling system, including performing, by an audio focus computing module and at a first time, a calibration and configuration of an audio focus model, performing, by the audio focus computing module at a second time, a steady-state monitoring of the information handling system, including applying the one or more configuration rules to perform one or more of the computer-implemented actions to automatically enable the video conference of the video conferencing application executing on the information handling system without user interaction.

The present disclosure relates to an information processing apparatus and an information processing method as well as a program that make it possible to control, by a partition provided on a boundary between two spaces, a visual shielding property and an auditory shielding property of a first space to a person in a second space in an interlocking relationship with each other in response to a distance between the person in the second space and the partition. A distance between the partition, which partitions the first space and the second space, and a person in the second space is measured, and transmittance of the partition and magnitude of output of audio in the first space to the second space are controlled in response to the measured distance. The present disclosure can be applied to a control apparatus for a partition section.

Systems and methods for suppressing noise and detecting voice input in a multi-channel audio signal captured by a plurality of microphones include (i) capturing a first audio signal via a first microphone and a second audio signal via a second microphone, wherein the first and second audio signals respectively comprises first and second noise content from a noise source; (ii) identifying the first noise content in the first audio signal; (iii) using the identified first noise content to determine an estimated noise content captured by the plurality of microphones; (iv) using the estimated noise content to suppress the first and second noise content in the first and second audio signals; (v) combining the suppressed first and second audio signals into a third audio signal; and (vi) determining that the third audio signal includes a voice input comprising a wake word.