Abstract: A hearing protection system is provided. The hearing protection system has a first active hearing protection system. The first active hearing protection system is level-dependent. The hearing protection system also has a second active hearing protection system. The second active hearing protection system is level-dependent. Each of the first and second active hearing protection systems are configured to operate in an independent mode and a dual-protection mode. A dual-protection mode level dependent function of the second active hearing protection system is different than an independent mode level dependent function.

Abstract: The present application discloses an audio device that has an inhibitory effect on sound waves emitted by a near-field sound source within a specified range and has an amplification effect on sound waves emitted from a far-field sound source outside the specified range. The audio device includes a first sound wave sensor to receive a sound wave and output a first signal based on the sound wave; a second sound wave sensor to receive the sound wave and output a second signal based on the sound wave; and a signal processing circuit coupled to the first sound wave sensor and the second sound wave sensor to generate an output signal based on the first signal and the second signal, wherein the audio device's near-field sensitivity to a sound wave is substantially lower than its far-field sensitivity to the sound wave.

Abstract: The embodiments of this application disclose an audio processing method and apparatus, an electronic device, and a storage medium. In the embodiments of this application, a current playback environment of audio may be obtained; audio recognition may be performed on ambient sound of the current playback environment in a case that the current playback environment is in a foreground state; foreground sound in the ambient sound may be determined according to an audio recognition result; the foreground sound in the ambient sound may be classified to determine a type of the foreground sound; and audio mixing may be performed on the foreground sound and the audio to obtain mixed playback sound based on the type of the foreground sound.

Abstract: Systems and methods for determining the incident angle of an acoustic wave are presented herein. One embodiment receives an acoustic wave at N transducers, where N is a natural number greater than or equal to 2; solves a set of N coupled differential equations modeling a set of N virtual coupled acoustic resonators using N coupled analog circuits, wherein each of the N coupled analog circuits receives an output signal from a unique one of the N transducers and outputs a voltage signal; and analyzes the N voltage signals output by the N coupled analog circuits to produce an estimate of the incident angle of the acoustic wave.

Abstract: A method performed by an electronic device in a room. The method performs an enrollment process in which a spatial profile of a location of an artificial sound source is created and performs an identification process that determines whether a sound event within the room is produced by the artificial sound source by 1) capturing the sound event using a microphone array and 2) determining a likelihood that the sound event occurred at the location of the artificial sound source.

Abstract: An audio system is described for calibrating one or more transducers for a user using the auditory steady state response (ASSR) of the user. The audio system presents an audio calibration signal to the user via a transducer array in the headset. The system measures electrical signals that are generated in the auditory cortex of the brain of the user in response to the presented audio calibration signal. The audio system determines an ASSR of the user based on the measured electrical signals. The audio system determines a value for one or more sound filter parameter based on the determined ASSR and a model. The audio system calibrates the transducer array using the determined sound filter parameters. The calibrated transducer array is used to present audio content to the user.

Abstract: An acoustic control system includes a peripheral wall provided so as to surround a seat, and an upper portion of the peripheral wall being opened, a speaker provided inside the peripheral wall and capable of outputting a reproduced sound from a sound source, a microphone provided toward an outside of the peripheral wall and configured to collect an environmental sound around the peripheral wall, and a processor. The processor configured to suppresse a level of the reproduced sound in a predetermined frequency band from the speaker based on the environmental sound collected by the microphone.

Abstract: A speaker comprises a housing, a transducer residing inside the housing, and at least one sound guiding hole located on the housing. The transducer generates vibrations. The vibrations produce a sound wave inside the housing and cause a leaked sound wave spreading outside the housing from a portion of the housing. The at least one sound guiding hole guides the sound wave inside the housing through the at least one sound guiding hole to an outside of the housing. The guided sound wave interferes with the leaked sound wave in a target region. The interference at a specific frequency relates to a distance between the at least one sound guiding hole and the portion of the housing.

Abstract: The present invention relates to a device and a computerized process for creating a virtual spatial audio representation of sound sources within a conference call application, which processes individual user voice or audio inputs according to a host or user selected spatial position, combining it with other users virtualized spatial audio positions, to simulate the sound properties of a real world in-person meeting, conference, convention, or event. This computerized process is embodied by software that can be used to process multiple incoming audio sources from multiple platform participants with different spatial audio filters to create a unique spatial positioning for each participant. It should be further noted that a single host could control the spatial positioning of all participants on the platform, or each participant could select from a defined set of available seats or positions based on their own personal preferences.

Abstract: The present disclosure discloses an acoustic output apparatus. The acoustic output apparatus may include at least one acoustic driver. The at least one acoustic driver may generate sound that is output through at least two sound guiding holes. Further, the acoustic output apparatus may include a supporting structure. The supporting structure may be configured to support the at least one acoustic driver. A baffle may be disposed between the at least two sound guiding holes. The baffle may increase an acoustic distance from at least one sound guiding hole of the at least two sound guiding holes to a user's ear.

Abstract: Disclosed herein are systems and methods for presenting mixed reality audio. In an example method, audio is presented to a user of a wearable head device. A first position of the user's head at a first time is determined based on one or more sensors of the wearable head device. A second position of the user's head at a second time later than the first time is determined based on the one or more sensors. An audio signal is determined based on a difference between the first position and the second position. The audio signal is presented to the user via a speaker of the wearable head device. Determining the audio signal comprises determining an origin of the audio signal in a virtual environment. Presenting the audio signal to the user comprises presenting the audio signal as if originating from the determined origin. Determining the origin of the audio signal comprises applying an offset to a position of the user's head.

Abstract: A technique for producing audio includes providing multiple audio tracks of respective sound sources of an audio performance and rendering a sound production of the audio performance at a listening venue by playing back the audio tracks on respective playback units.

Abstract: Presented herein are techniques for increasing sensitivity of a hearing prosthesis to sound signals received from the “side” of a recipient. The sensitivity of the hearing prosthesis to sound signals received from the side of a recipient is provided by a spatial pre-filter that is configured to use a primary reference signal (i.e., a first directional signal) and a side reference signal (i.e., a second directional signal having at least one null directed to the side of the recipient) to calculate a side gain mask. The side gain mask includes gains for each of a plurality of frequency channels associated with the received sound signals.

Abstract: The present disclosure relates to a signal processing device, a signal processing method, and a program that make it possible to readily achieve personalization of head-related transfer functions in all bands. A synthesis unit generates a third head-related transfer function by synthesizing a characteristic of a first band extracted from a first head-related transfer function of a user and a characteristic of a second band other than the first band extracted from a second head-related transfer function measured in a second measurement environment different from a first measurement environment in which the first head-related transfer function is measured. The present disclosure may be applied to, for example, a mobile terminal such as a smartphone.

Abstract: A hearing aid includes at least two microphones, providing respective at least two electric input signals representing sound around the user; a filter bank converting the electric input signals into signals as a function of time and frequency; a directional system connected to the microphones and being configured to provide a filtered signal in dependence of the electric input signals and fixed or adaptively updated beamformer weights. At least one direction to a target sound source is defined as a target direction. For each frequency band, one of the microphones is selected as a reference microphone, thereby providing a reference input signal for each frequency band. The reference microphone may be selected in dependence of directional data related to directional characteristics of the microphones. The reference microphone may be different for at least two frequency bands. The reference microphone may be adaptively selected based on a logic criterion.

Abstract: A method for calibrating an in-ear headphone to improve the frequency response heard by a user. The method including generating a sound signal and playing the sound signal at a driver when the in-ear headphone is placed within a user's ear canal, receiving a reflected sound signal at a first microphone, generating a frequency response based on the reflected sound signal, generating the user's ear drum response based on the frequency response, generating a second sound signal, modifying the second sound signal based on the user's ear drum response, and playing the modified second sound signal at the driver.

Abstract: A signal processing device includes a first panel displacement control unit to which a first audio signal is input, a second panel displacement control unit to which a second audio signal is input, and a control unit configured to control the first panel displacement control unit and the second panel displacement control unit. The first panel displacement control unit includes a first gain adjustment unit configured to adjust a level of the first audio signal, the second panel displacement control unit includes a second gain adjustment unit configured to adjust a level of the second audio signal, and the control unit includes a correlation determination unit configured to determine presence or absence of a correlation between the first audio signal and the second audio signal, and a gain control unit configured to control a level adjustment amount in each of the first gain adjustment unit and the second gain adjustment unit on the basis of a determination result of the correlation determination unit.

Abstract: A sound spatialisation method includes determining digital processing parameters to be applied to sound signals to be broadcast by a set of at least two loudspeakers in order to reproduce a virtual sound source at a desired position, and restoring sound signals by the loudspeakers during which the digital processing parameters are applied to the sound signals. The sound spatialisation method also includes defining a trajectory defined by a set of N points, with two consecutive points of the trajectory being connected together by a curve, and positioning during which the desired position of the virtual sound source is defined on the trajectory.

Abstract: A MEMS microphone includes a substrate, a diaphragm disposed over the substrate to cover the cavity, the diaphragm defining an air gap together with the back plate, and the diaphragm being spaced apart from the substrate, a back plate disposed over the diaphragm and in the vibration area, an upper insulation layer to cover the back plate, a plurality of chamber portions provided in the supporting area, a lower insulation layer provided under the upper insulation layer and on the substrate, and an intermediate insulation layer provided between the lower insulation layer and the upper insulation layer and disposed further from the vibration area than the chamber portions.

Abstract: The present disclosure provides an acoustic output apparatus. The acoustic output apparatus includes at least one low-frequency acoustic driver that outputs sounds from at least two first sound guiding holes, at least one high-frequency acoustic driver that outputs sounds from at least two second sound guiding holes, and a support component. The support component may be configured to support the at least one high-frequency acoustic driver and the at least one low-frequency acoustic driver, and cause the at least two first sound guiding holes and the at least two second sound guiding holes to locate away from a position of an ear of a user.

Abstract: A set of user pools can be determined based on location data associated with each device in an audio/video (A/V) conference. A key active user can be determined for each user pool of the set of user pools based on valid audio signals received from each device within each user pool. A determination can be made whether there is feedback within each user pool. Responsive to determining feedback in at least one user pool, speakers of devices within the at least one user pool can be disconnected except for the key active user device within each respective user pool.

Abstract: A method for spatial audio signal processing including: obtaining, from a first capture device, at least one first audio signal and at least one first direction parameter for at least one frequency band; obtaining, from a second capture device, at least one second audio signal and at least one second direction parameter for the at least one frequency band; obtaining a first position associated with the first capture device; obtaining a second position associated with the second capture device; determining a distance parameter for the at least one frequency band in relation to the first position based, at least partially, on the at least one first direction parameter and the at least one second direction parameter; and enabling an output and/or store of the at least one first audio signal, the at least one first direction parameter and the distance parameter.

Abstract: The present disclosure generally relates to user interfaces for managing spatial audio. Some exemplary techniques include user interfaces for transitioning between visual elements. Some exemplary techniques include user interfaces for previewing audio. Some exemplary techniques include user interfaces for discovering music. Some exemplary techniques include user interfaces for managing headphone transparency. Some exemplary techniques include user interfaces for manipulating multiple audio streams of an audio source.

Abstract: A first audio computing system to establish a connection with an audio source within a first network, establish a connection with a second audio computing system within a second network, receive, from the audio source via the connection within the first network, a first audio packet, the first audio packet including audio data, receive, from the second audio computing system, a receipt time, the receipt time indicating when the second audio computing system received, from the audio source within the first network, a second audio packet, the second audio packet including the audio data included in the first audio packet, calculate a rendering time based on a time of receiving the first audio packet and the receipt time, send the rendering time to the second audio computing system via the second network, and output audio based on the audio data.

Abstract: A method performed by an electronic device in a room. The method performs an enrollment process in which a spatial profile of a location of an artificial sound source is created and performs an identification process that determines whether a sound event within the room is produced by the artificial sound source by 1) capturing the sound event using a microphone array and 2) determining a likelihood that the sound event occurred at the location of the artificial sound source.

Abstract: A microphone, comprising at least two electrodes, spaced apart, configured to have a magnetic field within a space between the at least two electrodes; a conductive fiber, suspended between the at least two electrodes; in an air or fluid space subject to waves; wherein the conductive fiber has a radius and length such that a movement of at least a central portion of the conductive fiber approximates an oscillating movement of air or fluid surrounding the conductive fiber along an axis normal to the conductive fiber. An electrical signal is produced between two of the at least two electrodes, due to a movement of the conductive fiber within a magnetic field, due to viscous drag of the moving air or fluid surrounding the conductive fiber. The microphone may have a noise floor of less than 69 dBA using an amplifier having an input noise of 10 nV/?Hz.

Abstract: An apparatus including a first part, the first part having at least one camera configured to capture images; a second part having at least one microphone configured to capture at least one audio signal, wherein one of the first part or second part is able to move relative to the other part and the apparatus including circuitry configured to: determine a parameter associated with the move; generate at least one output audio signal based on the parameter associated with the move and the at least one audio signal.

Abstract: A portable speaker, such as a subwoofer, that is configured to pair via Bluetooth from any device capable of streaming Bluetooth audio and act as the “sink.” The portable speaker is also configured to pair via Bluetooth to other audio equipment capable of streaming Bluetooth audio and act as the “source.” The portable speaker is also preferably configured to synchronize the audio signal going to the portable speaker and the other unit playing music, such that the user will not hear delays between the two units playing the audio stream. Preferably, the portable speaker is configured to be portable in which the size and weight is small enough for a person to carry, and is portable in which it operates with rechargeable batteries inside the unit, thus not needing any other power source such as AC outlet, or automobile power system. Another embodiment omits the speaker and instead provides a line out for connection to an external speaker, such as a subwoofer.

Abstract: Disclosed herein are systems and methods for presenting mixed reality audio. In an example method, audio is presented to a user of a wearable head device. A first position of the user's head at a first time is determined based on one or more sensors of the wearable head device. A second position of the user's head at a second time later than the first time is determined based on the one or more sensors. An audio signal is determined based on a difference between the first position and the second position. The audio signal is presented to the user via a speaker of the wearable head device. Determining the audio signal comprises determining an origin of the audio signal in a virtual environment. Presenting the audio signal to the user comprises presenting the audio signal as if originating from the determined origin. Determining the origin of the audio signal comprises applying an offset to a position of the user's head.

Abstract: A method includes: generating a first and second shift signal by using a phase of a sound signal regarding an M-channel or a S-channel, the sound signal of the M-channel and the sound signal of the S-channel being obtained by using a mid-side microphone, the sound signal of the S-channel including a positive channel and a negative channel, the first shift signal being configured to reduce a phase difference caused by a difference between a sound arrival distance to the M-channel and a sound arrival distance to the positive channel of the S-channel, the second shift signal being configured to reduce a phase difference caused by a difference between the sound arrival distance to the M-channel and a sound arrival distance to the negative channel of the S-channel; and approximately converting the first or second shift signal into an L-channel signal and an R-channel signal of an XY-microphone.

Abstract: Provided are an artificial intelligence (AI) system using a machine learning algorithm such as deep learning and an application of the AI system. Provided is a method of converting sound data, the method including acquiring binaural sound data; converting the binaural sound data by using a pre-generated training network model, based on a parameter indicating a context at a time of acquiring the binaural sound data; and outputting the converted binaural sound data.

Abstract: Concentric rings of speakers in a room are activated in sequence to emit test sounds that are picked up by microphones placed in the ears of a listener in the middle of the room. The output of the microphones is used to generate plural HRTF spheres that are concentric, and that can be convolved with each other to render a multi-distance HRTF filter that can be configured to “move” the perceived distance and bearing of audio objects “inside” the outer sphere as desired, without being restricted to placing the audio objects at the outer edges of a single sphere. Any desired placement of an audio object between spheres is done by interpolating between respective coefficients of the closest spheres.

Abstract: Virtualizing speakers for a headphone set can include determining a location of a display. Locations of one or more virtual speakers can be assigned based on the location of the display. A first virtual speaker can be located at the display. A position of a head of a user can be tracked. Audio content can be spatialized with a spatial renderer to generate spatialized audio signals, based on the tracked position of the head and the locations of the virtual speakers. Other aspects are also described and claimed.

Abstract: A system is provided for actively mitigating noise caused by vibration of a thin vehicle front windshield in an occupant compartment of a vehicle. The system includes a pair of laser scanners configured to be mountable inside the vehicle occupant compartment. Each laser scanner is configurable to scan an associated initial portion of the windshield simultaneously with the other laser scanner, when the windshield is vibrating.

Abstract: An electronic apparatus is provided. The electronic apparatus includes a camera, a processor and a memory configured to store at least one instruction executable by the processor where and the processor is configured to input audio data to an artificial intelligence model corresponding to user information, and obtain output audio data from the artificial intelligence model, and the artificial intelligence model is a model learned based on first learning audio data obtained by recording a sound source with a first recording device, second learning audio data obtained by recording the sound source with a second recording device, and information on a recording device for obtaining the second learning audio data, and the second learning audio data is binaural audio data.

Abstract: Provided is a sound image reproduction device, sound image reproduction method, and sound image reproduction program that can support monaural sound sources and is capable of imparting directivity to virtual sound sources in a space.

Abstract: Upon an attempt to communicate through an audio input source in a voice session, a determination can be made whether an audio quality received from a user through the audio input source is below an audio quality threshold. In response to determining that the audio quality through the audio input source is below the audio quality threshold, a first voice input can be received from the user from a first position. A second voice input can be received from the user from a second position. A first volume of the first voice input from the first position can be compared to a second volume of the second voice input from the second position. Feedback can be provided to the user indicating whether the user is closer or farther from the audio input source based on the comparison.

Abstract: A method performed by a processing node (10), comprising the steps of: i. obtaining (11), from at least one communication device (100), audio data (12) associated with a sound and storing (13) the audio data (12) in the processing node (10), ii. Obtaining (15) an event designation (16) associated with the sound and storing (17) the event designation (16) in the processing node (10), iii. determining (19) a model (20) which associates the audio data (12) with the event designation (16) and storing the model (21), and iv. Providing (23) the model (20) to the communication device (100). A method performed by the communication device (100), as well as a processing node (10), a communication device (100), a system (1000) and computer programs for performing the methods are also described.

Abstract: A method for spatial audio signal processing, including determining at least one direction parameter for at least one frequency band based on microphone signals received from a microphone array; processing the determined at least one direction parameter to determine at least one distance parameter for the at least one frequency band; and enabling an output and/or store of the at least one distance parameter, at least one audio signal, and the at least one direction parameter.

Abstract: A sound processing method performs near-end side sound collection processing to collect a sound on a near-end side and generate a sound collection signal, near-end side filter processing to adjust the sound collection signal using transfer characteristics of a far-end side, and near-end side sound emission processing to emit the sound collection signal that has been adjusted, from a speaker on the near-end side.

Abstract: A spatial-audio recording system includes a spatial-audio recording device including a plurality of microphones, and a computing device. The computing device is configured to determine a plane-wave transfer function for the spatial-audio recording device based on a physical shape of the spatial-audio recording device and to expand the plane-wave transfer function to generate a spherical-harmonics transfer function corresponding to the plane-wave transfer function. The computing device is further configured to retrieve a plurality of signals captured by the microphones, determine spherical-harmonics coefficients for an audio signal based on the plurality of captured signals and the spherical-harmonics transfer function, and generate the audio signal based on the determined spherical-harmonics coefficients.

Abstract: The present invention provides for an apparatus, system, and method for generating a head related audio transfer function in real time. Specifically, the present invention utilizes unique structural components including a tragus structure and an antihelix structure in connection with a microphone in order to communicate the location of a sound in three-dimensional space to a user. The invention also utilizes an audio processor to digitally process the head related audio transfer function.

Abstract: An electronic device according to an embodiment may include: a first sound input device configured to obtain external sound and produce a first signal and a processor operatively connected to the first sound input device. The processor may be configured to: receive the first signal from the first sound input device; produce a first high-frequency signal by passing the first signal through a high-pass filter to; determine a first energy value of the first high-frequency signal; determine a second energy value of the first signal; compare a product of the second energy value of the first signal and the first energy value of the first high-frequency signal with a first threshold value to produce a first result; and determine whether the first sound input device is blocked based on the first result. In addition, various other embodiments may be provided.

Abstract: A method of detecting a replay attack on a voice biometrics system comprises: receiving an audio signal representing speech; detecting a magnetic field; determining if there is a correlation between the audio signal and the magnetic field; and if there is a correlation between the audio signal and the magnetic field, determining that the audio signal may result from a replay attack.

Abstract: A visible light audio system is operable to enable free space optical communication of audio signals via transmission of modulated light intensity at a light source to a photo diode being operably engaged with a demodulator and audio output device. Embodiments of the present disclosure enable a plurality of visible light transmitting apparatuses being installed in a commercial or residential dwelling and operably engaged over a network to combine their microphone inputs via spatial, amplitude, spectral, and/or temporal filtering and physical and geometrical modeling methods to separate one or more acoustic sources using a visible light audio system comprising an array of light sources being operable to receive an audio source input.

Abstract: A device to apply noise reduction to ambisonic signals includes a memory configured to store noise data corresponding to microphones in a microphone array. A processor is configured to perform signal processing operations on signals captured by microphones in the microphone array to generate multiple sets of ambisonic signals including a first set corresponding to a first particular ambisonic order and a second set corresponding to a second particular ambisonic order. The processor is configured to perform a first noise reduction operation that includes applying a first gain factor to each ambisonic signal in the first set and to perform a second noise reduction operation that includes applying a second gain factor to each ambisonic signal in the second set. The first gain factor and the second gain factor are based on the noise data, and the second gain factor is distinct from the first gain factor.

Abstract: Embodiments relate to a headset that filters sounds according to a direction of a gaze of a user wearing the headset. The user wears the headset including an eye tracking unit and one or more microphones. The eye tracking unit tracks an orientation of an eye of the user to determine the direction of the gaze of the user. The direction of the gaze may be different from a facing direction of the headset. According to the determined direction of the gaze of the user, input sound signals generated by the microphones can be beamformed to amplify or emphasize sound originating from the direction of the gaze.

Abstract: A microphone to generate A format signals used for ambisonics includes: a body of the microphone, first through fourth microphone elements provided facing sound pickup directions different from each other in the body and configured to output respective first signals to be components of the A format signals; and a six-axis sensor configured to detect displacement of the body and output information on a position of the body.

Abstract: A acoustic device includes a pinna replica, wherein an inner surface of the pinna replica faces the exterior so that exterior sounds are received by the pinna replica and a microphone unit for collecting said exterior sounds, said microphone unit being configured to emit a first electric signal comprising information on the collected exterior sounds.

Abstract: Apparatus for testing earphone apparatus (10) during manufacture includes a head simulator (30) including an ear simulator (40) defining a passageway (42) leading to an external opening (44), a and an eardrum microphone (46) mounted in the passageway (42) of the ear simulator (40). The apparatus (10) includes one or more of: an ear plate (50) for simulating an outer ear, the ear plate (50) forming part of the ear simulator (40) and defining a substantially planar earphone engagement surface (52) substantially encircling the external opening (44); a mounting guide system for assisting correct placement of earphone apparatus on the head simulator (30) relative to the ear simulator (40); and a test module (100) for performing rapid automated testing of earphone apparatus mounted on the head simulator (30).