Abstract: In some examples, a sensor node measures, using a thermometer, an ambient temperature of a location in which the wood item is located, measures, using a hygrometer, an ambient humidity of the location, measures, using an ohmmeter, a resistance of the wood item between two metal fasteners used to mount the sensor node to the wood item, determines, based on the resistance, a moisture content of the wood item, and determines, based on the ambient temperature, the ambient humidity, and the moisture content, whether conditions are conducive to pest activity. If conducive, the sensor node uses a microphone that is coupled to the wood item to capture audio data and determine whether the audio data indicates pest activity. If pest activity is detected, the sensor node sends a notification of pest activity, the notification includes the ambient temperature, the ambient humidity, the moisture content, and the audio data.

Abstract: A device can receive an audio signal and determine a measure of correlation between the audio signal and a microphone signal. The audio signal can be attenuated based on the measure of correlation. The audio signal can be used to drive one or more speakers of the device. Other aspects are described and claimed.

Abstract: The present invention relates to an active sound device utilizing audio recognition, the device comprising a data input unit for inputting audio data from an external terminal, an output unit for outputting the inputted audio data, an audio detection unit for detecting an audio signal generated from outside, a determination unit for determining the audio signal detected by the audio detection unit, and a control unit for, on the basis of information determined by the determination unit, controlling the output of the audio data inputted from the data input unit.

Abstract: An electronic eyewear device includes a display and a speaker system adapted to present augment reality objects and associated sounds in a scene being viewed by the user. A processor receives one or more audio tracks respectively associated with one or more augmented reality objects, encodes the audio tracks into an aggregated audio track including the audio tracks, a header for each audio track that uniquely identifies each respective audio track, and an aggregate header that identifies the number of tracks in the aggregated audio track. The processor transfers the aggregated audio track to an audio processor that uses the header for each audio track and the aggregate header to separate the audio tracks from the aggregated audio track. The audio processor processes the audio tracks independently in parallel and provides the audio tracks to the speaker system for presentation with the augmented reality objects.

Abstract: Methods, systems, computer-readable media, devices, and apparatuses for synchronized mode transitions are presented. A first device configured to be worn at an ear includes a processor configured to, in a first contextual mode, receive a first mode change request from a second device to transition at a first time to a second contextual mode. The first mode change request is based on detection of a trigger condition. The processor is further configured to, at the first time, transition from the first contextual mode to the second contextual mode based on the first mode change request.

Abstract: An active noise cancellation (ANC) method applied to a headset includes obtaining a first group of filtering parameters, and performing noise cancellation using the first group of filtering parameters. The first group of filtering parameters is one of N1 groups of filtering parameters prestored in the headset. The N1 groups of filtering parameters are respectively used to perform noise cancellation on ambient sound in N1 leakage states. The N1 leakage states are formed by the headset and N1 different ear canal environments. N1 is a positive integer greater than or equal to two.

Abstract: An example sound recognition method may include sampling input sound based on a preset sampling rate; performing Fast Fourier Transform (FFT) on the sampled input sound based on at least one of random FFT numbers or random hop lengths, and generating a two-dimensional (2D) feature map with a time axis and a frequency axis from the sampled input sound on which FFT is performed; training a neural network model, which recognizes sound, with a plurality of 2D feature maps including the first 2D feature map and an nth 2D feature map as training data.

Abstract: A reception device includes a first electronic component and a second electronic component. The first electronic component includes a first signal processor and a first control circuit controlling the first signal processor. The second electronic component includes a second signal processor and a second control circuit controlling the second signal processor. The first control circuit controls the first signal processor such that transmission of a reception signal from a first transmitter of the first signal processor to the second electronic component and transmission of an analog sound signal from a second transmitter of the first signal processor to the second electronic component are simultaneously started.

Abstract: Hearing instruments, such as hearing aids, may improve a quality of presented audio through the use of a binaural application, such as beamforming. The binaural application may require communication between the hearing instruments so that audio from a left hearing instrument may be combined with audio from a right hearing instrument. The combining at a hearing instrument can require synchronizing audio sampled locally with sampled audio received from wireless communication. This synchronization may cause a noticeable delay of an output of the binaural application if the latency of the wireless communication is not low (e.g., a few samples of delay). Presented herein is a low-latency communication protocol that communicates packets on a sample-by-sample basis and that compensates for delays caused by overhead protocol data transmitted with the audio data.

Abstract: Embodiments of this application provide an audio data processing method and apparatus, and a sound box system, which are related to the field of audio technologies, and improve sound quality of an audio playing device. The sound box system includes a full-frequency sound box and a low-frequency sound box. The full-frequency sound box is physically connected to the low-frequency sound box by using a first fastening part of the full-frequency sound box and a second fastening part of the low-frequency sound box. Furthermore, the full-frequency sound box communicates with the low-frequency sound box by using a first communication part of the first fastening part and a second communication part of the second fastening part, where the first fastening part and the second fastening part are a group of paired connection parts, and the first communication part and the second communication part are a group of paired communication parts.

Abstract: A method, apparatus and computer program are provided for assessing a continuity of audio service by comparing a previous audio service with a first spatial audio service that uses user head-tracking and a second audio service to identify which of the first or second spatial audio service provides a continuity of audio service with respect to the previous audio service. If the first or second spatial audio service is assessed to provide continuity of audio service, the respective spatial audio service is selectively enabled. The first spatial audio service controls or sets at least one directional property of at least one sound source and the first spatial audio service is assessed to provide continuity of audio service if it can use head-tracking to control or set the at least one directional property of at least one sound source.

Abstract: A sonic ranging method for a mobile phone and a wireless earphone. Firstly, the mobile phone performs an acoustic response test on the wireless earphone. The mobile phone generates a first audio sequence at a first time slot. The wireless earphone recognizes the first audio sequence to generate a second audio sequence. When the mobile phone recognizes the second audio sequence, the mobile phone estimates a length of elapsed time according to the time difference between a current time slot and the first time slot. When the mobile phone obtains the length of elapsed time, a distance between the wireless earphone and the mobile phone may be determined according to a delay compensation parameter and the length of elapsed time. Various potential applications can be implemented based on the invention without the needs for additional hardware circuits in the mobile phone.

Abstract: A bilateral hearing device system includes a hearing device. The hearing device includes a first input transducer configured for providing a first electric input signal, a second input transducer configured for providing a second electric input signal, a voice detector module configured to detect own voice of a user, the voice detector comprising a first band-pass filter configured for band-pass filtering the first electric input signal and a second band-pass filter configured for band-pass filtering the second electric input signal. The hearing device comprises a processing unit configured to provide a first electric output signal based on the first and second electric input signals; and an output transducer configured to provide an acoustic output signal, wherein the voice detector module is configured for notifying a detection of the own voice to the processing unit if one or more detection criteria are satisfied.

Abstract: The disclosure provides an approach for adaptive sound image width enhancement. A method of adaptively correcting sound image width is provided. The method includes obtaining an audio signal. The audio signal is associated with one or more audio channels. Each audio channel is associated with a position of an audio source with respect to a reference point within a local reproduction system. The method includes decomposing the audio signal into mid signal, side signal, and unprocessed signal components. The method includes applying respective correction gains to each of the mid signal, side signal, and unprocessed signal components. The respective correction gains are based on a physical distance between a first loudspeaker and a second loudspeaker within the local reproduction system. The method includes rendering, after applying the respective correction gains, the mid signal, side signal, and unprocessed signal components to the first and second loudspeakers.

Abstract: In at least one embodiment, a microphone assembly including a substrate, a printed circuit board (PCB), a micro-electro-mechanical systems (MEMS) transducer, a first lid, and a second lid is provided. The substrate defines a first port that extends completely therethrough. The PCB defines a sound opening that extends completely therethrough. The MEMS transducer is positioned on a first side of the substrate. The first lid defines a second port and covers the MEMS transducer and the first port. The first lid and the substrate define a front volume of air that surrounds the MEMS transducer. The second lid is positioned on the second side of the PCB. A cavity of the second lid, the sound opening of the PCB, the sound opening of the PCB, and the first port of the substrate define a back volume of air that is greater than the front volume of air.

Abstract: Provided is a method of reproducing an ambisonic signal in a virtual reality (VR) space. The ambisonic signal reproduction method may include receiving an ambisonic signal, mapping the ambisonic signal to channels localized on a sphere according to an equivalent spatial domain (ESD) standard corresponding to an order of the ambisonic signal, and performing a sound field reproduction in the VR space based on the channels localized on the sphere.

Abstract: A first-order differential microphone array (FODMA) with a steerable beamformer is constructed by specifying a target beampattern for the FODMA at a steering angle ? and then decomposing the target beampattern into a first sub-beampattern and a second sub-beampattern based on the steering angle ?. A first sub-beamformer and a second sub-beamformer are generated to each filter signals from microphones of the FODMA, wherein the first sub-beamformer is associated with the first sub-beampattern, and the second sub-beamformer is associated with the second sub-beampattern. The steerable beamformer is then generated based on the first sub-beamformer and the second sub-beamformer. The decomposing of the target beampattern into a first sub-beampattern and a second sub-beampattern includes dividing the target beampattern into a sum of a first-order cosine (cardioid) first sub-beampattern and a first-order sinusoidal (dipole) second sub-beampattern.

Abstract: A system configured to perform content recognition using fingerprinting to recognize known media content. A device determines fingerprints based on decoded content data to be sent using a media interface component to an output component. Metadata related to the content/device/fingerprint may also be created. The fingerprints and metadata are sent by the device to a supporting system for orchestration and matching of the fingerprints to known media content.

Abstract: In some example, a hub receives a notification of activity from a particular sensor node of a plurality of sensor nodes that are monitoring an area. The notification includes sensor data including: a difference in decibels between a first volume of a first filtered audio and a second volume of a second filtered audio and at least a portion of the second filtered audio. The hub performs an analysis of the sensor data using an artificial intelligence algorithm. The hub determines, based on the analysis, a presence of one or more tunneling activities and sends, by a communications interface of the hub, a message indicating the presence of the one or more tunneling activities to a computing device.

Abstract: A method of an electronic apparatus for audio signal processing includes obtaining a parameter related to spatialization of an audio object, obtaining rendering information based on the parameter related to spatialization, and rendering the audio object based on the rendering information. The parameter related to spatialization includes at least one of an object parameter of a feature of at least one of the audio object or a video object associated with the audio object, an electronic apparatus parameter of a feature of the electronic apparatus, or a user parameter of a feature of a user.