In one aspect, a first device includes at least one processor and storage accessible to the at least one processor. The storage includes instructions that may be executable by the processor to receive input from a camera and identify a second device based on the input from the camera. The second device may include at least one speaker and at least one microphone. The instructions may also be executable to identify a current location of the second device within an environment based on the input from the camera and to identify a current location of an object within the environment that is different from the second device. The instructions may then be executable to provide a command to alter operation of the at least one speaker and/or the at least one microphone based on the current location of the second device and the current location of the object.

Systems and methods for optimizing voice detection via a network microphone device are disclosed herein. In one example, individual microphones of a network microphone device detect sound. The sound data is captured in a first buffer and analyzed to detect a trigger event. Metadata associated with the sound data is captured in a second buffer and provided to at least one network device to determine at least one characteristic of the detected sound based on the metadata. The network device provides a response that includes an instruction, based on the determined characteristic, to modify at least one performance parameter of the NMD. The NMD then modifies the at least one performance parameter based on the instruction.

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.

A method and an apparatus for processing voice are provided. The method is applied to a decision-making device in communication with a distributed microphone array and the distributed microphone array comprises a plurality of sub-arrays. The method comprises: obtaining, for each sub-array, an awakening voice signal received by each microphone of the sub-array; determining, for each sub-array, a frequency domain signal corresponding to each awakening voice signal of the sub-array, and a first cross-correlation function between every two frequency domain signals; determining an awakened sub-array based on each first cross-correlation function for each sub-array.

A method and an electronic device for detecting crosstalk data are provided. The method for detecting crosstalk data can detect whether an audio data stream includes crosstalk data.

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.

A sound input/output device for a vehicle includes: sound collecting portions that are provided within a vehicle cabin and that collect voices of vehicle occupants; outputting portions that are provided within the vehicle cabin, and that output sound or images to respective seats; an awakeness degree judging section that judges degrees of awakeness of the vehicle occupants; and an output control section that, in a case in which it is judged, based on a voice collected by the sound collecting portion, that content relating to the vehicle or a vehicle periphery has been spoken, causes information relating to the vehicle or the vehicle periphery to be outputted from each outputting portion that corresponds to a seat in which a vehicle occupant, whose degree of awakeness is higher than a predetermined value, sits.

A system for managing moderator and audience audio captured by personal mobile devices and broadcast by a public address system in the same venue is disclosed. In various aspects, the system includes a wireless network hub. The wireless network hub transmits and receives wireless data from a plurality of personal mobile devices. A network manager component manages communications of the personal mobile devices with the wireless network hub. The network manager uses measured dynamic network parameters and determines conditions for maintaining voice stream quality according to configurable rules.

Traditional audio feedback elimination systems may attempt to reduce the effect of the audio feedback by simply scaling down the audio volume of the signal frequencies that are prone to howling. Other traditional feedback elimination systems may also employ adaptive notch filtering to detect and notch the so-called singing or howling frequencies as they occur in real-time. Such devices may typically have several knobs and buttons needing tuning, for example: the number of adaptive parametric equalizers (PEQs) versus fixed PEQs; attack and decay timers; and/or PEQ bandwidth. Rather than removing the singing frequencies with PEQs, the devices described herein attempt to holistically model the feedback audio and then remove the entire feedback signal. Two advantages of the devices described herein are: 1.) the system can operate at a much larger loop-gain (and hence with a much higher loudspeaker volume); and 2) setup is greatly simplified (i.e., no tuning knobs or buttons).

A method and a device of sustainably updating a coefficient vector of a finite impulse response FIRfilter. The method includes obtaining (21) a time-varying regularization factor used for iteratively updating the coefficient vector of the FIR filter in a case that the coefficient vector of the FIR filter is used for processing a preset signal; updating (22) the coefficient vector of the FIR filter according to the time-varying regularization factor.