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: 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: 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 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: 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: 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.

Abstract: An example may include detecting, via a controller, one or more microphones and one or more speakers in an area, measuring, via the one or more microphones, an initial frequency response of an audio signal generated by the one or more speakers inside the area and generating an initial room performance rating, comparing the initial frequency response to a target frequency response, creating audio compensation values to apply to the one or more speakers based on the comparison, and applying the audio compensation values to the one or more speakers.

Abstract: Methods, systems, computer-readable media, and apparatuses for manipulating a soundfield are presented. Some configurations include receiving a bitstream that comprises metadata and a soundfield description; parsing the metadata to obtain an effect identifier and at least one effect parameter value; and applying, to the soundfield description, an effect identified by the effect identifier. The applying may include using the at least one effect parameter value to apply the identified effect to the soundfield description.

Abstract: A 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 based on one or more detection criteria, 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 further 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 further configured for notifying a detection of the own voice to the processing unit if one or more of the detection criteria are satisfied.

Abstract: A hearing system comprises a hearing device, e.g. a hearing aid or a headset, configured to be worn by a user. The hearing device comprises at least one input transducer for providing at least one electric input signal representative of sound in the environment of the hearing device, wherein said at least one electric input signal comprises a target signal component assumed to be of current interest to the user, and a noise component. The hearing device further comprises a noise control system configured to provide an estimate of said target signal component and an estimate of said noise component and to apply a statistical structure to said noise component to thereby provide a modified noise component comprising said statistical structure; and to determine a modified estimate of said target signal component in dependence of said modified noise component. Thereby an improved segregation of sound sources may be provided.

Abstract: A sound pickup device damping structure and a sound pickup apparatus, the sound pickup device damping structure includes: a housing, including a side wall and a peripheral wall connected with the side wall; a circuit board, fixedly connected with the peripheral wall, wherein the circuit board and the side wall and the peripheral wall enclose to form an accommodating chamber; at least one sound pickup component, located in the accommodating chamber and electrically connected with the circuit board; and a first damping layer, disposed between the side wall and the sound pickup component; wherein a second sound pickup hole in the side wall, a hole channel in the first damping layer and a first sound pickup hole disposed in the sound pickup component are communicated with each other.

Abstract: An apparatus (303) comprising means configured to: obtain at least one audio signal (112); determine at least one direction parameter (506) of at least one propagating sound associated with the at least one audio signal (112); determine at least one direction (302) of at least one sound source associated with the at least one audio signal (112); modify the at least one direction parameter (506) in accordance with the determined at least one direction (302); and output the modified at least one direction parameter (508).

Abstract: Transducer steering and configuration systems and methods using a local positioning system are provided. The position and/or orientation of transducers, devices, and/or objects within a physical environment may be utilized to enable steering of lobes and nulls of the transducers, to create self-assembling arrays of the transducers, and to enable monitoring and configuration of the transducers, devices, and objects through an augmented reality interface. The transducers and devices may be more optimally configured which can result in better capture of sound, better reproduction of sound, improved system performance, and increased user satisfaction.

Abstract: There is provided a signal processing device that includes a processing unit configured to collect, through a microphone, a mixed sound signal as a mixed sound of a speaker reproduction signal and a target signal, input a first suppression signal resulting from suppression of the speaker reproduction signal from the mixed sound signal by linear processing, the speaker reproduction signal, and the mixed sound signal collected through the microphone, and output a second suppression signal resulting from further suppression of the speaker reproduction signal from the first suppression signal by non-linear processing.