Abstract: For online audio/video conferencing applications deployed in an open office environment, using shared conference devices, it can be advantageous to define an acoustic fence. A non-participant audio received from outside the acoustic fence can be considered noise and filtered out before transmission of an audio signal to a far end recipient. Three suppression stages are used to filter the non-participant audio. The first suppression stage uses beamformers for suppression. The second suppression stage is mask-based, and the third suppression stage is reference-based. The three suppression stages filter out non-participant audio signals, having a wide range of frequencies.

Abstract: A method for providing navigation assistance using a head-worn device that has a camera, several microphones, and several speakers. The method captures sound in an environment as a plurality of microphone audio signals, captures, using the camera, a scene of the environment as a digital image, and processes the digital image to detect an object therein. The method selects, in response to the detection of the object, one of several navigation audio rendering modes. The several navigation audio rendering modes include a first mode that activates an acoustic transparency function to cause the speakers to reproduce the sound of the environment, a second mode that sonifies the object and activates the acoustic transparency function, and a third mode that sonifies the object, partially activates the acoustic transparency function, and an activates active noise cancellation function.

Abstract: The invention discloses a stereo enhancement system and a stereo enhancement method. The stereo enhancement system includes a beamforming unit and a signal processing unit. The beamforming unit is used for receiving a plurality of input sound signals and generating a plurality of beamforming sound signals corresponding to a plurality of direction intervals respectively. The signal processing unit is coupled to the beamforming unit and used for receiving the plurality of beamforming sound signals corresponding to the plurality of direction intervals respectively and generating a first synthesized output sound signal and a second synthesized sound signal accordingly.

Abstract: Presented herein are techniques to enhance the audio portion of a video conference. In one embodiment, a method includes determining, using a multi-microphone array, a direction of arrival of sound signals from a user, detecting, using an image from a camera, a face of the user, determining a position of the face of the user with respect to a position of the camera, and forming a spatial beam for the multi-microphone array based on the direction of arrival of sound signals from the user and the position of the face of the user.

Abstract: Various systems and methods are provided for conducting acoustical measurements via an inflatable mannequin. In one embodiment, a system comprises an inflatable mannequin that comprises a detachable mannequin head, a plurality of inflatable anatomical parts, and a plurality of configurable joints, an inflation system that calibrates and controls inflation of the plurality of anatomical parts of the inflatable mannequin; and a joint control system that calibrates and controls the plurality of configurable joints via joint motors. In another embodiment, a first method comprises selecting a mannequin profile based on sizes of anatomical parts of the inflatable mannequin and second method comprises conducting acoustical measurements with the selected mannequin profile. In this way, acoustical measurements may be conducted to develop audio processing schemes that widen the sweet spot in an interior of a vehicle.

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: According to an example embodiment, an electronic device may include: a communicator comprising communication circuitry configured to establish communication with a pair of earphones, and a processor configured to: determine whether each of the earphones is being worn and initiate recording of a sound input through a microphone of an earphone determined to be in a worn state among the earphones.

Abstract: The disclosure herein generally relates to capturing, acoustic pre-processing, encoding, decoding, and rendering of directional audio of an audio scene. In particular, it relates to a device adapted to modify a directional property of a captured directional audio in response to spatial data of a microphone system capturing the directional audio. The disclosure further relates to a rendering device configured to modify a directional property of a received directional audio in response to received spatial data.

Abstract: The present disclosure relates to an acoustic output apparatus. The acoustic output apparatus comprising: at least one low-frequency acoustic driver that outputs sound from at least two first sound guiding holes; at least one high-frequency acoustic driver that outputs sound from at least two second sound guiding holes; and a controller configured to cause the low-frequency acoustic driver to output sound in a first frequency range, and cause the high-frequency acoustic driver to output sound in a second frequency range, wherein the second frequency range includes frequencies higher than the first frequency range.

Abstract: Various techniques pertaining to non-coherent noise reduction for audio enhancement on a multi-microphone mobile device are proposed. A processor receives a plurality of signals from a plurality of audio sensors corresponding to a plurality of channels responsive to sensing by the plurality of audio sensors. The processor then performs a non-coherent noise reduction on one or more signals of the plurality of signals to suppress one or more non-coherent noises in each of the one or more signals based on a respective signal-to-noise ratio (SNR) associated with each of the one or more signals. The processor further combines the plurality of signals subsequent the noise reduction to generate an output signal.

Abstract: In one implementation, a method of changing an audio property of an object is performed at a device including one or more processors coupled to non-transitory memory. The method includes displaying, using a display, a representation of a scene including a representation of an object associated with an audio property. The method includes displaying, using the display, in association with the representation of the object, a manipulator indicating a value of the audio property. The method includes receiving, using one or more input devices, a user input interacting with the manipulator. The method includes, in response to receiving the user input, changing the value of the audio property based on the user input and displaying, using the display, the manipulator indicating the changed value of the audio property.

Abstract: A method (800) to detect a hotword in a spoken utterance (120) includes receiving a sequence of input frames (210) characterizing streaming multi-channel audio (118). Each channel (119) of the streaming multi-channel audio includes respective audio features (510) captured by a separate dedicated microphone (107). For each input frame, the method includes processing, using a three-dimensional (3D) single value decomposition filter (SVDF) input layer (302) of a memorized neural network (300), the respective audio features of each channel in parallel and generating a corresponding multi-channel audio feature representation (420) based on a concatenation of the respective audio features (344). The method also includes generating, using sequentially-stacked SVDF layers (350), a probability score (360) indicating a presence of a hotword in the audio. The method also includes determining whether the probability score satisfies a threshold and, when satisfied, initiating a wake-up process on a user device (102).

Abstract: The present disclosure relates to a page presentation method, a display system, and a storage medium. The method includes presenting, on a page, a map comprising one or more virtual objects used to identify one or more corresponding electronic signs; and binding a sign content to be presented to the electronic signs corresponding to the one or more virtual objects.

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 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: 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: 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: 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: 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 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: 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: 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: 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: 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: 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.