A method for canceling an echo and an electronic device thereof are provided. The electronic device includes a housing, a communication module, a first speaker disposed in a first region of the housing, a second speaker disposed in a second region of the housing, a first microphone disposed adjacent to the first region, a second microphone disposed adjacent to the second region, and a processor. The processor is configured to receive a first audio signal from an external electronic device, and output a first given frequency band, and output a second given frequency band, and provide a first signal by applying a filter capable of passing a band corresponding to the second given frequency band, and provide a second signal by applying a filter capable of passing a band corresponding to the first given frequency band, and provide a second audio signal corresponding to the external audio signal, and transmit the second audio signal to the external electronic device.

Voice responsive in-wall devices are provided. In one example implementation, a power switch includes a housing mountable on or at least partially within a surface. The housing can have a front panel. The power switch can include an interface element disposed on the front panel and operable to receive a user input. The power switch can include a power interrupter operable to control power delivery to the powered load based at least in part on interaction with the interface element. The power switch can include one or more microphones operable to obtain audio input. The power switch can include one or more speakers configured to provide audio output. The power switch can include a communications interface operable to communicate data associated with the audio input over a communication link.

Receivers are arranged to detect a corresponding observed phase shift, observed attenuation or other observed signal parameters for its respective microphone. An electronic data processor is adapted to estimate a distribution or quantity of material on the sieve based on the observed phase shift, the observed attenuation or the other observed signal parameters relative to a reference phase shift, a reference attenuation or other reference signal parameter. The operator can be alerted via a user interface if the material on the sieve is unevenly distributed or matches a preestablished distribution profile, or the sieve can be adjusted by an actuator to promote a generally uniform distribution.

A method of and system for visualizing a sound source is disclosed. The method may include analyzing an audio signal received by a sound transducer to determine a positional direction of the sound source, determining whether the positional direction of the sound source falls outside a field of view of a user, and in response to determining that the positional direction of the sound source falls outside the field of view of the user, rendering on a display unit a visual representation of the sound source. The visual representation of the source is rendered on a virtual surface at a location within the field of view of the user, the location corresponding to at least one of a distance of the source from the user and a positional direction of the source with respect to the user.

A wearable device may include an image sensor configured to capture a plurality of images from an environment, a microphone configured to capture sounds from the environment, and at least one processor. The at least one processor may be programmed to receive audio signals representative of the sounds captured by the at least one microphone, and receive a first image including a representation of a first individual from among the plurality of images captured by the image sensor. The at least one processor may also be programmed to obtain a first audio segment from the audio signals using the first image. The first audio segment may include a first portion of the audio signals in which the first individual is speaking. The at least one processor may also be programmed to receive a second image including a representation of a second individual from among the plurality of images captured by the image sensor, and obtain a second audio segment from the audio signals using the second image. The second audio segment may include a second portion of the audio signals in which the second individual is speaking. The at least one processor may also be programmed to receive a third image including a representation of the first individual from among the plurality of images captured by the image sensor, and using the third image, obtain a third audio segment from the audio signals. The audio segment may include a third portion of the audio signals in which the first individual is speaking. The at least one processor may also associate the first and third audio segments with the first individual and associate the second audio segment with the second individual.

An audio renderer can have a machine learning model that jointly processes audio and visual information of an audiovisual recording. The audio renderer can generate output audio channels. Sounds captured in the audiovisual recording and present in the output audio channels are spatially mapped based on the joint processing of the audio and visual information by the machine learning model. Other aspects are described.

A noise cancellation system for an ear-mountable playback device having a speaker, a feedforward microphone and an error microphone comprises a filter chain for coupling the feedforward microphone to the speaker, the filter chain comprising a series connection or parallel connection of a coarse filter and a fine filter, and a noise control processor. The fine filter is formed of a set of sub-filters having a predefined frequency range, wherein the predefined frequency range of each of the sub-filters together forms an effective overall frequency range of the fine filter. The noise control processor is configured to calculate an error signal based on a first noise signal sensed by the feedforward microphone and on a second noise signal sensed by the error microphone, to perform an adaptation of coarse filter parameters of the coarse filter based on the error signal, and to perform a limited adaptation of fine filter parameters of each of the sub-filters based on the error signal, wherein limits of the limited adaptation comprise the predefined frequency ranges of the sub-filters and at least one of a gain limit and a Q factor limit.

An acoustic crosstalk suppression device includes a speaker estimation unit configured to estimate a main speaker based on voice signals collected by n units of microphones corresponding to n number of persons (n: an integer equal to or larger than 3); n units of filter update units each of which is configured to update a parameter of a filter configured to generate a suppression signal of a crosstalk component included in a voice signal of the main speaker; and a crosstalk suppression unit configured to suppress the crosstalk component by using a synthesis suppression signal generated by the maximum (n-1) units of filter update units corresponding to reference signals collected by the maximum (n-1) units of microphones.

A method, including an action of receiving first data based on data based on ambient sound captured with a first microphone, and further including an action of receiving second data based on data based on the ambient sound captured with a second microphone, wherein the first microphone is a part of a hearing prosthesis, the second microphone is part of an indoor sound capture system or indoor sound capture sub-system, and the method further comprises comparing the first data to the second data.

Microphone arrays (MAs) are described that position and vent microphones so that performance of a noise suppression system coupled to the microphone array is enhanced. The MA includes at least two physical microphones to receive acoustic signals. The physical microphones make use of a common rear vent (actual or virtual) that samples a common pressure source. The MA includes a physical directional microphone configuration and a virtual directional microphone configuration. By making the input to the rear vents of the microphones (actual or virtual) as similar as possible, the real-world filter to be modeled becomes much simpler to model using an adaptive filter.