Abstract: A microelectromechanical system (MEMS) motor includes a substrate, a backplate, and a diaphragm. The substrate has a first surface and a second surface. The second surface has a slot that extends at least partially into the substrate. A port extends through the substrate. The backplate is mounted to the first surface of the substrate, and the backplate covers at least a portion of the port. The diaphragm is between the backplate and the substrate. The diaphragm moves with respect to the backplate in response to acoustic energy that passes through the port.

Abstract: A computer system detects an occurrence of a respective event. In response, and in accordance with a determination that a first wearable audio output component of a wearable audio output device is at least partially in a first ear of the user and that a second wearable audio output component of the wearable audio output device is at least partially in a second ear of the user, the computer system displays acoustic seal information for the wearable audio output device, including concurrently displaying, via the display device: a first indication of a quality of a first acoustic seal between the first wearable audio output component and the first ear of the user; and a second indication, distinct from the first indication, of a quality of a second acoustic seal between the second wearable audio output component and the second ear of the user.

Abstract: One or more earbuds comprise one or more external connectors for removably coupling the earbuds with an additional article. The one or more earbuds are additionally able to comprise magnets for removably coupling with each other. In some embodiment, the one or more earbuds are configured to removably couple with a base unit. Based on a coupling and decoupling of the one or more earbuds with the base unit, a signal is sent to control a remotely located electronic device.

Abstract: The apparatus and methods of the present invention provide improved accuracy of response for a tactile transducer included in a body-mounted device such as a headphone, VR/AR headset or similar device. Accuracy is increased through the application of digital signal processing, such as with Infinite Impulse Response filters or Finite Impulse Response filters.

Abstract: A sound synthesis system is provided with a loudspeaker to project sound indicative of synthesized motor sound in response to receiving a synthesized sound (SS) signal, and a processor. The processor is programmed to: estimate motor sound based on a sensor signal indicative of sound present within a passenger compartment; identify a dominant motor harmonic of the motor sound with an amplitude and a frequency; determine an enrichment value of the motor sound; determine if the motor sound is unenriched based on a comparison of the enrichment value to an enrichment threshold value; generate at least one additional motor harmonic with a first frequency that is different than the frequency of the dominant motor harmonic in response to the motor sound being unenriched; and provide the SS signal to the loudspeaker, wherein the SS signal is indicative of the at least one additional motor harmonic.

Abstract: Provided is an information processing device that controls and presents sound information in an appropriate form to a user who acts in an environment on the basis of situation recognition including recognition of the environment and recognition of the actions of the user.

Abstract: Determination of a composite acoustic parameter value for a headset is presented herein. A directionally enhanced audio signal is generated based on audio signals from an acoustic sensor array and a spatial signal enhancement filter that is directed for enhancement of a sound source. A SNR improvement value is determined based on a SNR value of the directionally enhanced audio signal and a SNR value of an audio signal from an acoustic sensor of the acoustic sensor array. The SNR improvement value is input into a model that maps SNR improvement values to spatial acoustic parameters to determine a spatial acoustic parameter. A temporal acoustic parameter is determined based on the audio signals. The composite acoustic parameter value is determined based on the spatial acoustic parameter and a temporal acoustic parameter value. Audio content presented to a user is adjusted based in part on the composite acoustic parameter value.

Abstract: A hearing device includes: a first microphone and a second microphone for provision of a first microphone input signal and a second microphone input signal, respectively; a voice detector module configured to process the first microphone input signal and the second microphone input signal, the voice detector module configured to detect own-voice of a user of the hearing device; a processor configured to process the first microphone input signal and the second microphone input signal for provision of an electrical output signal based on the first microphone input signal and the second microphone input signal; and a receiver configured to convert the electrical output signal to an audio output signal; wherein the voice detector module is configured to notify a detection of the own-voice to the processor if at least two of a first voice criterion, a second voice criterion, and a third voice criterion are satisfied.

Abstract: A sound quality improvement based on artificial intelligence is disclosed. A sound control method based on artificial intelligence according to an embodiment of the present disclosure provides a different call sound quality for each person based on a plurality of person information stored in an address book of a mobile terminal. The mobile terminal and 5G network of the present disclosure may be associated with an artificial intelligence module, a drone ((Unmanned Aerial Vehicle, UAV), a robot, an AR (Augmented Reality) device, a VR (Virtual Reality) device, a device associated with 5G services, etc.

Abstract: An electronic device and method that automatically adjusts an audio output volume level based on a live environmental acoustic scenario input via a microphone using a machine learning algorithm trained with Human Activity Recognition (HAR). Equipped with such an intelligence the electronic device classifies ambient sounds occurring in the environment of the listening area in which the device is situated into different acoustic scenario mappings such a voice or conversation, for an ambient human conversation detected event, and noise, such as for example a vacuum cleaner or dish washer noise detected event, and automatically adjust the audio output volume accordingly.

Abstract: The amount of far-field noise transmitted by a primary communication device in an open-plan office environment is reduced by defining an acoustic perimeter of reference microphones around the primary device. Reference microphones generate a reference audio input including far-field noise in the proximity of the primary device. The primary device generates a main audio input including the voice of the primary speaker as well as background noise. Reference audio input is compared to main audio input to identify the background noise portion of the main audio signal. A noise reduction algorithm suppresses the identified background noise in the main audio signal. The one or more reference microphones defining the acoustic perimeter may be included in separate microphone devices placed in proximity to the main desktop phone, microphones within other nearby desktop telephone devices, or a combination of both types of devices.

Abstract: The Bluetooth device (DEV) controlling a plurality of wireless audio devices, comprises: a wireless communication circuit, to receive a wireless audio device identifier from each wireless audio device; a sound processing circuit to apply independent audio effects on a main audio stream such that the sound processing circuit outputs as many modified audio streams as the wireless communication circuit has received identifiers of wireless audio devices, on the basis of the characteristics of each wireless audio device; an allocation circuit to allocate each modified audio stream to a respective wireless audio device, a transmission circuit to wirelessly transmit through a Bluetooth usage each modified audio stream to said respective wireless audio device for emission by the wireless audio devices.

Abstract: A hearing device includes: a wireless communication unit configured for receiving an antenna signal carrying information in a first frequency band having a first centre frequency, and information in a second frequency band having a second centre frequency; wherein the wireless communication unit comprises: a local oscillator configured to provide a local oscillator signal; a mixer configured to receive the local oscillator signal and mixing the antenna signal with the local oscillator signal, the mixer providing an output signal; and an IF filter configured to receive the output signal; wherein an IF centre frequency associated with the IF filter is equal to a fraction of a difference between the second centre frequency and the first centre frequency; and wherein a local oscillator frequency associated with the local oscillator is equal to a difference between the second centre frequency and the IF centre frequency.

Abstract: A method operates a hearing device. The hearing device has a microphone by which ambient sound is picked up and is converted into an input signal that has a wanted component and a noise component. A stationarity of the input signal is determined. A signal-to-noise ratio of the input signal is determined on a basis of a scaling factor. The scaling factor is determined on a basis of the stationarity, namely on a basis of a function that indicates the scaling factor on a basis of the stationarity of the input signal. A corresponding hearing device implements such a method.

Abstract: Personal audio systems and methods are disclosed. A personal audio system includes a voice activity detector to determine whether or not an ambient audio stream contains voice activity, a pitch estimator to determine a frequency of a fundamental component of an annoyance noise contained in the ambient audio stream, and a filter bank to attenuate the fundamental component and at least one harmonic component of the annoyance noise to generate a personal audio stream. The filter bank implements a first filter function when the ambient audio stream does not contain voice activity, or a second filter function when the ambient audio stream contains voice activity.

Abstract: An electronic device includes one or more microphones that generate audio signals and a wind noise detection subsystem. The electronic device may also include a wind noise reduction subsystem. The wind noise detection subsystem applies multiple wind noise detection techniques to the set of audio signals to generate corresponding indications of whether wind noise is present. The wind noise detection subsystem determines whether wind noise is present based on the indications generated by each detection technique and generates an overall indication of whether wind noise is present. The wind noise reduction subsystem applies one or more wind noise reduction techniques to the audio signal if wind noise is detected. The wind noise detection and reduction techniques may work in multiple domains (e.g., the time, spatial, and frequency domains).

Abstract: An audio signal processing device includes a memory; and a processor configured to execute receiving a command to determine one set of parameters from among parameters calculated according to positions at which a sound is localized, the positions being set relative to positions of audio output devices; and processing a sound to be output from the audio output devices, by using the one set of parameters determined based on the received command.

Abstract: The present disclosure discloses a method and an apparatus for testing a speaker, an electronic device and a storage medium. A specific implementation includes: obtaining first audio data recorded by a microphone integrated with the speaker in ambient white noise; analyzing the first audio data to derive a first analysis result; and determining whether there is a defect in the microphone according to the first analysis result. Hence, these allow for testing a completed set on an assembled speaker to ensure the consistency of a microphone test and improve the accuracy of the test result.

Abstract: This application relates to audio driving circuitry (100), and in particular to audio driving circuitry for outputting first and second audio driving signals for driving a stereo audio load (106), which may be a stereo audio load of an accessory apparatus (102) removably coupled to the audio driving circuitry in use. A load monitor (111) is provided for monitoring to monitor, from a monitoring node (112), an indication of a common mode return current passing through a common return path, together with an indication of a common mode component of the first and second audio driving signals and to determine an impedance characteristic of the stereo audio load. The load monitor (111) can provide dynamic monitoring of any significant change in load impedance. In some embodiments the load monitor (111) comprises an adaptive filter (301) which adapts a parameter of the filter which is related to the load impedance so as to determine the indication of load impedance.

Abstract: A system method provides for multi-port wind noise protection. A sound may include a desired component such as speech and an undesired component such as wind. Multiple apertures on a housing receive the sound and conduct it to a microphone. The undesired component such as wind is uncorrelated at the apertures and mixes at the microphone, attenuating in amplitude while the desired component such as speech is correlated at the apertures. In this manner, the signal to noise ratio between the desired component and undesired component is improved at the microphone.