A method and apparatus to assist people with hearing loss. An augmented reality device with microphones and a display captures speech of a person talking to the wearer of the device and displays real-time captions in the wearer's field of view, while optionally not captioning the wearer's own speech. The microphone system in this apparatus inverts the use of microphones in augmented reality devices by analyzing and processing environmental sounds while ignoring the wearer's own voice.

Provided is a signal processing device including a main speech detection unit configured to detect, by using a neural network, whether or not a signal input to a sound collection device assigned to each of at least two speakers includes a main speech that is a voice of the corresponding speaker, and output frame information indicating presence or absence of the main speech.

A microphone includes: a first connection port for connecting to an external device; a second connection port for connecting to a second microphone via a cable; and a determination part that determines to be in a master mode, in which electric power is supplied through the second connection port to the second microphone connected via the cable, when the cable is connected to the second connection port in a state where electric power is supplied from the external device through the first connection port, and determines to be in a slave mode, in which electric power is supplied through the second connection port from the second microphone connected via the cable, when the cable is connected to the second connection port in a state where no electric power is supplied through the first connection port.

A voice controlled assistant has a housing to hold one or more microphones, one or more speakers, and various computing components. The housing has an elongated cylindrical body extending along a center axis between a base end and a top end. The microphone(s) are mounted in the top end and the speaker(s) are mounted proximal to the base end. The microphone(s) and speaker(s) are coaxially aligned along the center axis. The speaker(s) are oriented to output sound directionally toward the base end and opposite to the microphone(s) in the top end. The sound may then be redirected in a radial outward direction from the center axis at the base end so that the sound is output symmetric to, and equidistance from, the microphone(s).

A device for estimating Direction of Arrival (DOA) of sound from Q≥1 sound sources is provided. The device is configured to obtain a phase difference matrix, which includes measured phase difference values, each of the measured phase difference values being a measured value of a phase difference between two microphone units for a frequency bin in a range of frequencies of the sound. The device is further configured to generate a replicated phase difference matrix by replicating the measured phase difference values to other potential sinusoidal periods, calculate a DOA value for each phase difference value in the replicated phase difference matrix, and determine, as Q DOA results, the Q most prominent peak values in a histogram generated based on the calculated DOA values.

An apparatus including at least one processor and at least one memory including computer code for one or more programs, the at least one memory and the computer code configured, with the at least one processor, to cause the apparatus at least to: generate content lock information for a content lock, wherein the content lock information enables control of audio signal processing associated with audio signals related to one or more audio sources based on a position and/or orientation input.

Proposed is a method for determining, by a movable robot, a position of a speaker, wherein the movable robot includes first to fourth microphones installed at four vertexes of a quadrangle of a horizontal cross section of the robot respectively, wherein the method includes: receiving a wake-up voice through first and third microphones disposed respectively at first and third vertices in a diagonal direction; obtaining a first reference value of the first microphone and a second reference value of the third microphone based on the received wake-up voice; comparing the obtained first and second reference values to select the first microphone; selecting a second microphone disposed at a second vertex, wherein the first and second microphones are on a front side of the quadrangle; calculating a sound source localization (SSL) value based on the selected first and second microphones; and tracking a position of the speaker based on the SSL value.

An ambient sound environment is captured by a microphone array of an autonomous vehicle traveling in the ambient sound environment. A perception module of the autonomous vehicle classifies sounds and localizes sound sources in the ambient sound environment. Classification is performed using spectrum analysis and/or machine learning. In an embodiment, sound sources within a field of view (FOV) of an image sensor of the autonomous vehicle are localized in a visual scene generated by the perception module. In an embodiment, one or more sound sources outside the FOV of the image sensors are localized in a static digital map. Localization is performed using parametric or non-parametric techniques and/or machine learning. The output of the perception module is input into a planning module of the autonomous vehicle to plan a route or trajectory for the autonomous vehicle in the ambient sound environment.

A device has a microphone array that acquires sound data and a camera that acquires image data. A portion of the device may be moveable by one or more actuators. Responsive to the user, the portion of the device is moved toward an estimated direction of the user. The estimated direction is based on sensor data including the sound data and the image data. First variance values for individual sound direction values are calculated. Data derived from the image data or data from other sensors may be used to modify the first variance values and determine second data comprising second variances. The second data may be processed to determine the estimated direction of the user. For example, the second data may be processed by both a forward and a backward Kalman filter, and the output combined to determine an estimated direction toward the user.

The present disclosure provides an acoustic device including a microphone array, a processor, and at least one speaker. The microphone array may be configured to acquire an environmental noise. The processor may be configured to estimate a sound field at a target spatial position using the microphone array. The target spatial position may be closer to an ear canal of a user than each microphone in the microphone array. The processor may be configured to generate a noise reduction signal based on the environmental noise and the sound field estimation of the target spatial position. The at least one speaker may be configured to output a target signal based on the noise reduction signal. The target signal may be used to reduce the environmental noise. The microphone array may be arranged in a target area to minimize an interference signal from the at least one speaker to the microphone array.