Systems and methods for audio output control are disclosed. Audio may be output via a speaker of a communal device associated with a first portion of an environment. A user may provide a user utterance indicating an intent to add another device in a second portion of the environment to the audio-output session, and/or an intent to move the audio-output session from the first device to the second device, and/or an intent to remove a device from an audio-output session. Based on this determined intent, audio-session queues may be associated and dissociated from devices and device states may be altered to effectuate the intent of the user utterance.

A mixing device of a first signal and a second signal on a time-frequency plane, includes a control signal generation unit to generate a control signal indicating whether to perform prioritized mixing that includes amplification of the first signal and attenuation of the second signal; and a gain derivation unit to derive a first gain for amplifying the first signal and a second gain for attenuating the second signal based on the control signal. The control signal takes at least a first value and a second value different from the first value, wherein the first value is not continuous beyond a predetermined bandwidth on a frequency axis. The mixing device applies the prioritized mixing to the first and second signals when the control signal indicates the first value, and applies simple addition to the first and second signals when the control signal indicates the second value.

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.

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.

A conversation support system supports conversation of passengers in an automobile. Seats of at least two or more rows are placed in the automobile, and the conversation support system includes a speech switching control part configured to designate a desired sound pickup and reproduce area based on designation by a first passenger sitting in a first seat, a first target speech emphasizing part configured to output a signal obtained by emphasizing speech emitted from the designated sound pickup and reproduce area to a loud speaker corresponding to the first seat, and a second target speech emphasizing part configured to output the signal obtained by emphasizing speech, which is collected with a microphone corresponding to the first seat and which is emitted from the first seat, to a loud speaker corresponding to the sound pickup and reproduce area.

In one aspect, a first playback device is configured to (i) receive a set of voice signals, (ii) process the set of voice signals using a first set of audio processing algorithms, (iii) identify, from the set of voice signals, at least two voice signals that are to be further processed, (iv) determine that the first playback device does not have a threshold amount of computational power available, (v) receive an indication of an available amount of computational power of a second playback device, (vi) send the at least two voice signals to the second playback device, (vii) cause the second playback device to process the at least two voice signals using a second set of audio processing algorithms, (viii) receive, from the second playback device, the processed at least two voice signals, and (ix) combine the processed at least two voice signals into a combined voice signal.

A method, apparatus and computer program, the method including enabling an output of an audio mixer to be rendered for a user where the user is located within a sound space, wherein at least one input channel is provided to the audio mixer and the at least one input channel receives a plurality of microphone output signals obtained by a plurality of microphones recording the sound space; determining that a first microphone records one or more sound objects within the sound space; and in response to the determining, enabling one or more of the plurality of microphone output signals to be, at least partially, removed from the at least one input channel to the audio mixer.

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.

Systems, apparatus, and methods for processing audio signals associated with conferencing devices communicatively connected in a daisy-chain configuration using local connection ports included on each device are provided. One method involving a first conferencing device comprises receiving auxiliary mixed microphone signal(s) from at least one other conferencing device via at least one local connection port, each auxiliary signal comprising a mix of microphone signals captured by the at least one other conferencing device; determining a gain adjustment value for each auxiliary mixed microphone signal based on a daisy-chain position of the at least one other conferencing device relative to the position of the first conferencing device; adjusting a gain value for each auxiliary mixed microphone signal based on the corresponding gain adjustment value; generating a loudspeaker output signal from the gain-adjusted auxiliary mixed microphone signal(s); and providing the loudspeaker signal to the loudspeaker of the first conferencing device.

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.