Described herein is a system for recalibrating an audio configuration for mobile or moving devices. The system may configure a multi-device output group to generate synchronous output audio using multiple devices. For example, the output group may include a first device generating a first portion of output audio corresponding to a first channel and a second device generating a second portion of the output audio corresponding to a second channel. If the second device detects motion and/or movement indicating a change in its location, the system may recalibrate the output group to continue generating the output audio without the second device. For example, the first device or a new device can generate the second portion of the output audio instead of the second device. When the second device returns, the system can recalibrate the output group to include the second device again.

Sensing Method and Sensor System A sensing method comprises using a vertical cavity surface emitting laser (VCSEL) to oscillate and emit a laser beam. A diaphragm is used to reflect a portion of the laser beam back into the VCSEL. This method can be referred as self mixing interferometry. A current or voltage at the VCSEL is monitored, and is used to sense movement of the diaphragm. This allows a property external to the VCSEL to be sensed without using a photo-detector.

An electronic device is provided. The electronic device includes a speaker, a microphone, a processor, and a memory. For example, the electronic device may obtain a first audio signal during a first specified time by using the microphone, may identify reference noise intensity based on the first audio signal, may obtain a second audio signal exceeding the reference noise intensity by using the microphone, may turn off the microphone based on a fact that a third audio signal having the reference noise intensity or less is obtained during a second specified time or longer, and may modulate and output the second audio signal through the speaker while the microphone is turned off.

Provided is an electronic device for controlling surface heart and a method of controlling the electronic device. The electronic device includes a speaker, a temperature sensor, a memory, and a processor electrically coupled to the speaker, the temperature sensor, and the memory. The processor obtains first temperature information based on impedance information measured in a coil included in the speaker; obtains second temperature information measured by the temperature sensor, the second temperature information based on a heat source disposed adjacent to the speaker; predicts a surface temperature of a surface area of the electronic device, opposite to an internal area in which the speaker is disposed, based on the first temperature information and the second temperature information using a nonlinear approximation function; and controls an audio signal input to the speaker based on the predicted surface temperature.

A method for estimating an acoustic influence of walls of a room, comprising emitting a known excitation sound signal, receiving a set of measurement signals, each measurement signal being received by one microphone in a microphone array and each measurement signal including a set of echoes caused by reflections by the walls, solving a linear system of equations to identify locations of image source and estimating the acoustic influence based these image sources. The signal model includes a convolution of: the excitation signal, a multichannel filter (M) representing the relative delays of the microphones in the microphone array, the relative delays determined based on a known geometry of the microphone array, and a directivity model ν(n, p) of the driver(s) in the form of an anechoic far-field impulse response as a function of transmit angle.

A fitting agent for a hearing device and related method is disclosed, wherein the fitting agent is configured to initialize a user model comprising a user preference function; obtain a primary test setting for the hearing device; obtain a secondary test setting for the hearing device; present the primary test setting and the secondary test setting to a user; detect a user input of a preferred test setting indicative of a preference for either the primary test setting or the secondary test setting; and update the user model based on hearing device parameters of the preferred test setting, wherein to obtain the secondary test setting comprises: obtain a candidate set of candidate test settings; determine an uncertainty parameter for each candidate test setting; and select the secondary test setting from the candidate set of candidate test settings based on the uncertainty parameters of the candidate test settings.

Systems and methods are provided for providing media content based on playback zone awareness. In one aspect, a computing system receives, via a network interface, zone data from the media playback system, wherein the zone data includes an indication of a particular zone of the media playback system, and wherein the particular zone comprises at least one playback device. The computing system identifies audio content based on (i) the indication of the particular zone and (ii) contextual data associated with the particular zone, and provides, via the network interface, an indication of the identified audio content to the media playback system.

A management server includes a memory, and at least one processor configured to control an audio client system, by execution of instructions stored in the memory. The at least one processor is configured to acquire a first transfer function measured for a portion of or for all of the first signal path in the audio client system. The at least one processor is also configured to generate a virtual second signal path for a portion of or for all of the first signal path in the audio client system, and calculate a second transfer function for a portion of or for all of the generated virtual second signal path. The at least one processor is also configured to determine a condition of the audio client system, based on a result of a comparison between the first transfer function and the second transfer function.

Techniques of performing acoustic echo cancellation involve providing a bi-magnitude filtering operation that performs a first filtering operation when a magnitude of an incoming audio signal to be output from a loudspeaker is less than a specified threshold and a second filtering operation when the magnitude of the incoming audio signal is greater than the threshold. The first filtering operation may take the form of a convolution between the incoming audio signal and a first impulse response function. The second filtering operation may take the form of a convolution between a nonlinear function of the incoming audio signal and a second impulse response function. For such a convolution, the bi-magnitude filtering operation involves providing, as the incoming audio signal, samples of the incoming audio signal over a specified window of time. The first and second impulse response functions may be determined from an input signal input into a microphone.

A device for calculating loudspeaker signals for a plurality of loudspeakers while using a plurality of audio sources, an audio source including an audio signal, includes a forward transform stage for transforming each audio signal, block-by-block, to a spectral domain so as to obtain for each audio signal a plurality of temporally consecutive short-term spectra, a memory for storing a plurality of temporally consecutive short-term spectra for each audio signal, a memory access controller for accessing a specific short-term spectrum among the plurality of short-term spectra for a combination consisting of a loudspeaker and an audio signal on the basis of a delay value, a filter stage for filtering the specific short-term spectrum for the combination of the audio signal and the loudspeaker by using a filter provided for the combination of the audio signal and the loudspeaker, so that a filtered shot-term spectrum is obtained for each combination of an audio signal and a loudspeaker, a summing stage for summing up the filtered short-term spectra for a loudspeaker so as to obtain summed-up short-term spectra for each loudspeaker, and a backtransform stage for backtransforming, block-by-block, summed-up short-term spectra for the loudspeakers to a time domain so as to obtain the loudspeaker signals.