Abstract: A method for controlling a motion tracking system, including the steps of: digitally processing sound waves detected by a plurality of microphones so as to detect a voice of a user and estimate a first direction of the user; digitally processing electromagnetic waves captured by antennas so as to detect data packets transmitted to a computing apparatus by sensors and estimate second directions of each sensor; digitally averaging the second directions so as to provide an average direction for the sensors; digitally computing a difference between the first direction and the average direction; and starting to digitally track motion of the user based on measurements of each sensor when the computed difference does not exceed a predetermined difference threshold.

Abstract: An audio system with a loudspeaker configured to create an audio output, a sensor configured to detect at least the presence of at least one person at a position relative to the loudspeaker, and a processor configured to cause the loudspeaker to alter the audio output based on the position of the at least one person relative to the loudspeaker. Altering of the audio output includes adjusting at least one of a volume, a time delay, an equalization, or a radiation pattern of the audio output.

Abstract: A beamformer, for a differential microphone array (DMA) including a number M of microphones, is constructed based on a specified target directivity factor (DF) value for the DMA. An N order beampattern is generated for the DMA, wherein N is an integer and a first DF value corresponding to the N order beampattern is greater than the target DF value. An N?1 order beampattern is generated for the DMA, wherein a second DF value corresponding to the N?1 order beampattern is greater than the target DF value. A fractional order beampattern is generated for the DMA, wherein a third DF value corresponding to the fractional order beampattern matches the target DF value and the fractional order beampattern comprises a first fractional contribution from the N order beampattern and a second fractional contribution from the N?1 order beampattern.

Abstract: A communications system (300) comprising: an antenna (320) that comprises a plurality of serially connected sub-antenna elements (322); and a signal generator (324) configured to provide a transmission signal to the antenna (320) for propagating along the sub-antenna elements (322). The transmission signal comprises a plurality of serial symbol packets. The signal generator (324) is configured to set the phase of the serial symbol packets such that when they align with predefined ones of the sub-antenna elements (322) the antenna (322) provides a beamformed signal.

Abstract: A method, apparatus and computer program is described comprising: generating or obtaining at least one augmented reality image for presentation to a user, the augmented reality images comprising one or more virtual objects, displays or devices of an augmented reality system; attenuating real-world audio of the augmented reality system; and controlling the provision of audio to the user in at least one audio focus or beamform direction relating to at least one selected real-world object of the augmented reality system.

Abstract: An ear-worn device may include two or more microphones configured to generate time-domain audio signals, each of the two or more microphones configured to generate one of the time-domain audio signals; processing circuitry comprising analog processing circuitry, digital processing circuitry, beamforming circuitry, and short-time Fourier transformation (STFT) circuitry, the processing circuitry configured to generate, from the time-domain audio signals, one or more frequency-domain non-beamformed audio signals and one or more frequency-domain beamformed signals; and enhancement circuitry comprising neural network circuitry configured to receive multiple frequency-domain input audio signals originating from the one or more frequency-domain non-beamformed audio signals and the one or more frequency-domain beamformed signals, and implement a single neural network trained to generate, based on the multiple frequency-domain input audio signals, a noise-reduced and spatially-focused output audio signal or an output

Abstract: An audio device can sense sound in a physical environment using a plurality of microphones to generate a plurality of microphone signals. Clean speech can be extracted from microphone signals. Ambience can be extracted from the microphone signals. The clean speech can be encoded at a first compression level. The ambience can be encoded at a second compression level that is higher than the first compression level. Other aspects are also described and claimed.

Abstract: A binaural hearing system comprises first and second hearing aids, each comprising antenna and transceiver circuitry allowing the exchange of audio signals between them and a BTE-part adapted for being located at or behind the external ear (pinna) of the user and comprising front and rear input transducers providing respective front and rear electric input signals. Each of the hearing aids comprises primary and secondary adaptive 2-channel beamformers each providing a spatially filtered signal based on first and second beamformer-input signals. The primary and secondary 2-channel beamformers are coupled in a cascaded structure. The inputs to the primary 2-channel beamformers are, locally generated, front and rear electric input signals. The inputs to the secondary 2-channel beamformer may be beamformed signals from the first and second hearing aids respectively. The spatially filtered signal of the secondary 2-channel beamformer may comprise an estimate of a target signal in the environment of the user.

Abstract: The present disclosure relates to a method of processing audio content including directivity information for at least one sound source, the directivity information comprising a first set of first directivity unit vectors representing directivity directions and associated first directivity gains. The disclosure further relates to corresponding methods of encoding and decoding audio content including directivity information for at least one sound source.

Abstract: Various implementations include approaches for sound enhancement in far-field pickup. Certain implementations include a method of sound enhancement for a system including microphones for far-field pick up. The method can include: generating, using at least two microphones, a primary beam focused on a previously unknown desired signal look direction, the primary beam producing a primary signal configured to enhance the desired signal; generating, using at least two microphones, a reference beam focused on the desired signal look direction, the reference beam producing a reference signal configured to reject the desired signal; and removing, using at least one processor, components that correlate to the reference signal from the primary signal.

Abstract: An apparatus including circuitry configured to: provide a plurality of metrics associated with multiple directions, wherein the metrics are dependent, for the respective multiple directions, on an amount of direct-sound propagated; accumulate, for the respective multiple directions, values dependent upon the metrics associated with the respective multiple directions to enable determination of sound source directions.

Abstract: A metamaterial comprising, a plurality of acoustic vector field sensors, each configured to sense an acoustic vector field of a fluid within a fluid-filled space in response to fluid waves, and producing an electrical signal corresponding to the sensed acoustic vector field; a processor configured to perform a time and space transform on the electrical signal; and at least one phased array transducer, configured to emit fluid waves according to a produced acoustic vector field pattern dependent on a result of the time and space transform, a within a portion of the fluid.

Abstract: An apparatus comprising: an input configured to receive at least one audio signal input from at least one microphone; at least one microphone configuration determiner configured to provide for the at least one microphone a location on the apparatus; at least one sensor configured to provide at least one orientation of the apparatus; a recording mode determiner configured to determine at least one recording mode for the apparatus based on the location of the at least one microphone and the at least one orientation of the apparatus; a recording mode controller configured to determine at least one recording parameter for the at least one audio signal input from the at least one microphone based in the at feast one recording mode; and digital signal processor configured to apply the at least one recording parameter to the at least one audio signal input.

Abstract: The present disclosure relates to microphones and electronic devices having the same. The microphone may include a housing for receiving sound signals, at least two transducers for vibrating to generate electrical signals in response to the sound signals, and a processing circuit for processing the electrical signals. Each of the at least two transducers may provide a distinctive resonance peak to the microphone.

Abstract: An influence of hands holding an input device or fingers manipulating operation members on sound collection by a microphone is reduced. An input device (10) has a right held portion (10R) that has an upper surface on which an operation button (11) to be operated by a finger in the right hand is disposed, a left held portion (10L) that has an upper surface on which a direction key (12) to be operated by a finger in the left hand is disposed, and a middle portion (10M) positioned between the right held portion (10R) and the left held portion (10L). A first microphone (21) and a second microphone (22) are arranged in the middle portion (10M). The first microphone (21) is positioned rearward of a center of the middle portion (10M) in a front-rear direction.

Abstract: A spherically steerable microphone array structure is provided. The spherically steerable microphone array structure uses pressure and acoustic particle velocity signals obtained from sensors positioned co-planarly on a circular arc. The spherically steerable microphone array structure allows a calculation of all spatial partial derivatives of a sound field up to a given order. The spatial partial derivatives are used to obtain a spherical harmonic decomposition of a recorded sound field. Spherical harmonic decomposition coefficients are used in a spherically direction-invariant acoustic mode beamforming.

Abstract: A system for capturing sound comprising a plurality of discrete microphones (112, 14, 116, 118) and a processing system (408). The plurality of discrete microphones are arranged in a circular array. The processing system (408) arranged to perform a first signal processing algorithm on sound originating from one or more of a first set of directions relative to the array to isolate a first sound source. The processing system (408) is further arranged to perform a second signal processing algorithm on sound originating from one or more of a second set of directions relative to the array to isolate a second sound source therein. A method for receiving sound at a plurality of discrete microphones (112, 114, 116, 118) arranged in a circular array is also described.

Abstract: An antenna module is an array antenna in which an array of antenna elements extends in at least a first direction. The array of antenna elements in the first direction includes a first antenna group in a middle portion and a second antenna group in two end portions adjacent to the middle portion. The antenna elements in the first antenna group are unequally spaced. The spacing between adjacent antenna elements in the second antenna group is greater than the maximum spacing between adjacent elements in the first antenna group. The amplitude distribution in the antenna module as a whole in the first direction is in a unimodal form in which the amplitude of a radio-frequency signal fed to the antenna elements in the second antenna group is smaller than the amplitude of a radio-frequency signal fed to the antenna elements in the first antenna group.

Abstract: Embodiments include a planar microphone array comprising a first linear array arranged along a first axis; and a second linear array arranged along a second axis orthogonal to the first axis, a center of the second linear array aligned with a center of the first linear array, wherein each of the first linear array and the second linear array comprises a corresponding first set of microphone elements nested within a corresponding second set of microphone elements, and each set of microphone elements is arranged symmetrically about the center of the corresponding linear array, such that the first linear array and the second linear array are configured to generate a steerable directional polar pattern, the microphone elements of each linear array configured to capture audio signals. Embodiments also include a microphone system comprising the same and a method performed by processor(s) to generate an output signal for the same.

Abstract: A method and system for compensating a frequency response of a microphone array are provided. The method includes: multiple microphones in a microphone array respectively receiving a compensation signal sent from a calibration speaker and outputting multiple output signals respectively; determining a uniform frequency response of the multiple microphones based on the multiple output signals; calculating a compensation gain for each of the multiple microphones according to the uniform frequency response; and storing the calculated compensation gain of each microphone.

Abstract: Examples of the disclosure describe systems and methods for estimating acoustic properties of an environment. In an example method, a first audio signal is received via a microphone of a wearable head device. An envelope of the first audio signal is determined, and a first reverberation time is estimated based on the envelope of the first audio signal. A difference between the first reverberation time and a second reverberation time is determined. A change in the environment is determined based on the difference between the first reverberation time and the second reverberation time. A second audio signal is presented via a speaker of a wearable head device, wherein the second audio signal is based on the second reverberation time.

Abstract: A gaze direction of a user is determined by an eye-tracking system of a head mounted device. Audio data generated by at least one microphone is captured. Gaze-guided audio is generated from the audio data based on the gaze direction of the user.

Abstract: Systems and methods are provided for reducing unwanted noise in an electronic audio signal, wherein a computing device having a microphone is configured to receive signals from a sensor on an external device such as a camera, second microphone, or movement sensor. The signals from the sensor are used to identify sound information or characteristics of sounds made by a source of noise, and the audio signal of the microphone is modified to reduce unwanted sounds based on that sound information or based on sounds identified a second audio signal obtained by the second microphone, thereby improving teleconference and video conference audio quality and removing distracting noises from transmitted audio output.

Abstract: A conference device and a computer-implemented method for training a neural network are disclosed, the conference device comprising a conference controller; a microphone array comprising a plurality of microphones for provision of audio signals representing audio from one or more sound sources; a direction estimator connected to the conference controller and the microphone array, the direction estimator configured to obtain, from the microphone array, a plurality of audio signals including a first audio signal and a second audio signal; determine direction data based on the plurality of audio signals, the direction data comprising an indication of an estimated probability of voice activity for one or more directions, wherein to determine direction data comprises to apply an offline-trained neural network; and output audio data based on the direction data to the conference controller.

Abstract: Endfire linear array microphone systems and methods with consistent directionality and performance at different frequency ranges are provided. The endfire linear array microphone includes a delay and sum beamformer and a differential beamformer. The delay and sum beamformer may produce pickup patterns with good directionality at higher frequency ranges, but cause the pickup patterns to become more omnidirectional at lower frequencies. The differential beamformer may produce pickup patterns with good directionality at lower frequencies. By combining the delay and sum beamformer and differential beamformer within the linear array microphone, the overall directionality of the linear array microphone may be maintained at different frequency ranges while using the same microphone elements.

Abstract: Methods and systems may incorporate voice interaction and other audio interaction to facilitate access to prescription related information and processes. Particularly, voice/audio interactions may be utilized to achieve authentication to access prescription-related information and action capabilities. Additionally, voice/audio interactions may be utilized in performance of processes such as obtaining prescription refills and receiving reminders to consume prescription products.

Abstract: A microphone array system or microphone array unit for a conference system is provided that includes a front board, side walls and a plurality of microphone capsules arranged in or on the front board mountable on or in a ceiling of a conference room. The microphone array system or unit is adapted for generating a steerable beam within a maximum detection angle range. The microphone array system or microphone array unit includes a processing unit which is configured to receive the output signals of the microphone capsules and to steer the beam based on the received output signal of the microphone array. The processing unit is configured to control the microphone array to limit the detection angle range to exclude at least one predetermined exclusion sector in which a noise source is located.

Abstract: An example method of operation may include receiving audio data, via one or more microphones of a network device, from an audio source, determining a location of the audio data based on a direction and amplitude of the received audio data, modifying the audio data for output via a loudspeaker of the network device, and outputting, via the loudspeaker, the modified audio data.

Abstract: Embodiments include an array microphone comprising a plurality of microphone sets arranged in a linear pattern relative to a first axis and configured to cover a plurality of frequency bands. Each microphone set comprises a first microphone arranged along the first axis and a second microphone arranged along a second axis orthogonal to the first microphone, wherein a distance between adjacent microphones along the first axis is selected from a first group consisting of whole number multiples of a first value, and within each element, a distance between the first and second microphones along the second axis is selected from a second group consisting of whole number multiples of a second value.

Abstract: A hearing device includes: a first housing; a second housing; first and second primary microphones for provision of first and second primary microphone input signals; a secondary microphone in the second housing for provision of a secondary microphone input signal; a mixing module for provision of a mixer output based on a primary mixer input and a secondary mixer input, wherein the primary mixer input is based on the first primary microphone input signal and the second primary microphone input signal, and the secondary mixer input is based on the secondary microphone input signal; a processing unit configured to process the mixer output and to provide an electrical output signal; and a receiver for converting the electrical output signal to an audio output signal; wherein the mixing module is configured to mix a first component of the primary mixer input and a first component of the secondary mixer input.

Abstract: Provided herein are methods of activating immune response and/or treating cancer in a patient comprising administering to the patient a Gal3:TIM-3 inhibitor that interferes with the interaction between Gal3 and TIM-3, where said inhibitor is administered in an amount sufficient to activate immune response. Also provided are a humanized anti-Gal3 antibodies that can block the interaction between Gal3 and TIM3 and methods of using the anti-Gal3 antibody to treat cancer. Methods for determining if a patient's cancer is suitable for treatment with a Gal3:TIM-3 inhibitor and methods for selecting compounds that can block interaction between Gal3 and TIM-3, activating immune response and/or treating cancer are also provided.

Abstract: An audio signal processing method includes receiving an audio signal corresponding to a voice of a talker, obtaining an image of the talker, estimating position information of the talker using the image of the talker, generating, according to the estimated position information, a correction filter configured to compensate for an attenuation of the voice of the talker, performing filter processing on the audio signal using the generated correction filter, and outputting the audio signal on which the filter processing has been performed.

Abstract: A method for noise filtering including receiving directional data corresponding to the relative directions of at least one noise source and at least one target audio source; capturing noise data from the at least one noise source; capturing target audio data from the at least one target audio source; using the directional data to filter the noise data from the target audio data; and outputting filtered target audio.

Abstract: A light emission display device according to an aspect of the present invention includes: a light guide member that converts rays of incident light from a back face thereof to produce diffused light and emits the diffused light from the front face; a plurality of point light sources that are arranged at intervals on a side of the back face of the light guide member, in which the light guide member includes a light shield structure that partitions the front face into a plurality of light emission regions each corresponding to at least one point light source of the plurality of point light sources immediately below the front face, such that transmission of the rays of light to an adjacent light emission region is reduced.

Abstract: Disclosed are methods and systems which convert a multi-microphone input signal to a multichannel output signal making use of a time- and frequency-varying matrix. For each time and frequency tile, the matrix is derived as a function of a dominant direction of arrival and a steering strength parameter. Likewise, the dominant direction and steering strength parameter are derived from characteristics of the multi-microphone signals, where those characteristics include values representative of the inter-channel amplitude and group-delay differences.

Abstract: The invention relates to a variable-directivity MEMS microphone. The microphone comprises an acoustic cavity. The following components are provided inside the acoustic cavity: a first acoustic transducer for detecting an acoustic signal and converting the acoustic signal into a first acoustic conversion signal; a first pre-amplifier, connected to the first acoustic transducer, and configured for outputting a first electric signal; a second acoustic transducer for detecting an acoustic signal and converting the acoustic signal into a second acoustic conversion signal; a second pre-amplifier, connected to the second acoustic transducer, and configured for outputting a second electric signal; and a signal processing chip, connected to the first pre-amplifier and the second pre-amplifier, and configured for generating a directional output signal by performing an arithmetic operation on the first electric signal and the second electric signal under the action of a switching control signal.

Abstract: Cochlear implant systems can include a cochlear electrode, a stimulator in electrical communication with the cochlear electrode, a sensor configured to receive a stimulus signal and generate an input signal based on the received stimulus signal, and a signal processor in communication with the stimulator and the sensor. The signal processor can include an analog filtering stage configured to generate an analog filtered signal from a received input signal and a digital filtering stage configured to generate a digitally filtered signal from the analog filtered signal. The analog filtering stage and digital filtering stage can be used to normalize the frequency response of the digitally filtered signal with respect to the stimulus signal.

Abstract: An example method of operation may include designating sub-regions which collectively provide a defined reception space, receiving audio signals at a controller from the microphone arrays in the defined reception space, configuring the controller with known locations of each of the microphone arrays, assigning each of the sub-regions to at least one of the microphone arrays based on the known locations, and creating beamform tracking configurations for each of the microphone arrays based on their assigned sub-regions.

Abstract: A feed forward equalizer including a first set of filter taps having a first set of filter tap coefficients to be adapted and a second set of one or more filter taps having one or more filter tap coefficients to be constrained. The feed forward equalizer includes an adaptation component to determine a set of adapted filter tap coefficient values corresponding to the first set of filter tap coefficients and a constraint function component to determine a constrained filter tap coefficient value for the second set of the one or more filter taps having the one or more filter tap coefficients to be constrained using a constraint function based on at least a portion of the set of adapted filter tap coefficient values. The feed forward equalizer generates, based at least in part on the constrained filter tap coefficient value, an equalized signal including a set of estimated symbol values.

Abstract: A head-worn audio device is provided with a circuit for voice signal enhancement. The circuit comprises at least a plurality of microphones, arranged at predefined positions, where each microphone provides a microphone signal. The circuit further comprises a directivity pre-processor and a blind source separation processor. The directivity pre-processor is connected with the plurality of microphones to receive the microphone signals and being configured to provide at least a voice signal and a noise signal. Directivity pre-processing increases the mutual independence of the signals provided to the blind source separation processor and thus improves processing by blind source separation. The blind source separation processor receives at least the voice signal and the noise signal, and is configured to conduct blind source separation on at least the voice signal and the noise signal to provide at least an enhanced voice signal with reduced noise components.

Abstract: A vehicle driving control apparatus includes a communication interface configured to receive, from a sound sensor, a signal corresponding to sound that is generated in an external environment, and a processor configured to identify a sound object generating the sound, by obtaining a type of the sound object and either one or both of a direction of the sound object and a distance from the sound object to a vehicle including the vehicle driving control apparatus, based on the received signal, and control driving of the vehicle, based on the identified sound object.

Abstract: An audio processing apparatus includes a first microphone configured to obtain ambient sound, a second microphone configured to obtain noise from a noise source, a first conversion unit configured to perform a Fourier transform on an audio signal from the first microphone to generate a first audio signal, a second conversion unit configured to perform a Fourier transform on an audio signal from the second microphone to generate a second audio signal, a generation unit configured to generate noise data by using the second audio signal and a parameter related to the noise from the noise source, a subtraction unit configured to subtract the noise data from the first audio signal, and a third conversion unit configured to perform an inverse Fourier transform on an audio signal from the subtraction unit.

Abstract: Methods, apparatus, systems and articles of manufacture are disclosed to improve detection of audio signatures.

Abstract: A microphone module comprises a housing, an audio bus, and a first plurality of microphones in communication with the audio bus. The microphone module further comprises a module processor in communication with the first plurality of microphones and the audio bus. The module processor is configured to detect the presence of an array processor in communication with the audio bus, detect the presence of a second microphone module in communication with the audio bus, and configure the audio bus to pass audio signals from both the first plurality of microphones and the second microphone module to the array processor.

Abstract: A system is described herein. The system includes at least one hardware processor that is configured to identify a pre-determined acoustic barrier filter, wherein the acoustic barrier filter coincides with the physical acoustic barrier and receive an audio signal within a time window at the first microphone and the second microphone. The hardware processor is also configured to calculate a first measure of variability, a second measure of variability, a third measure of variability, and a fourth measure of variability. The hardware processor further concatenates the first measure of variability, the second measure of variability, the third measure of variability, and the fourth measure of variability to form a feature vector, and inputs the feature vector into a location classifier to obtain an audio source location.

Abstract: The embodiments of the present application provide an earphone abnormality processing method, an earphone, a system, and a storage medium. In the embodiments of the present application, an abnormal state self-detection function is added to an earphone of an existing dual-microphone, and when the earphone is in a self-detection state, acquiring a sound signal picked up from a specified sound source by a primary microphone and a secondary microphone; and determining the type of abnormal state of the earphone according to a frequency response curve of the sound signal picked up by the primary microphone and the secondary microphone, and further performing abnormality processing on the earphone by using a processing manner adapted to the abnormal state of the earphone, thereby solving the difficulty in the prior art of being unable to process the sound pickup abnormality of an earphone, and improving the usage performance of an earphone and also facilitating the prolonging of the service life of the earphone.

Abstract: A microphone array is described for use in ultra-high acoustical noise environments. The microphone array includes two directional close-talk microphones. The two microphones are separated by a short distance so that one microphone picks up more speech than the other. The microphone array can be used along with an adaptive noise removal program to remove a significant portion of noise from a speech signal of interest.

Abstract: Acoustic Voice Activity Detection (AVAD) methods and systems are described. The AVAD methods and systems, including corresponding algorithms or programs, use microphones to generate virtual directional microphones which have very similar noise responses and very dissimilar speech responses. The ratio of the energies of the virtual microphones is then calculated over a given window size and the ratio can then be used with a variety of methods to generate a VAD signal. The virtual microphones can be constructed using either an adaptive or a fixed filter.

Abstract: The disclosed device may include an optical structure configured to house various sensors directed toward a user. The sensors may be configured to gather data through at least one layer of the optical structure. The device may also a vent bracket positioned between the optical structure and an outer covering of the device. The vent bracket may be positioned to provide an opening between the optical structure and the vent bracket, allowing air to flow through the opening. The device may also include various microphones positioned in recessed ports between the optical structure and the vent bracket. These openings may allow external sounds to reach the microphones in the device. Various other methods of manufacturing, systems, and computer-readable media are also disclosed.

Abstract: A network device for an intercom network for duplex audio communication between users of the intercom network has a housing and a communication module in or at the housing and that can establish a radio link for transmitting audio signals to and receiving audio signals from another user. The radio link is subject to the DECT protocol or another protocol, and the audio signals to the communication module is broken by reflection into a plurality of radio paths. Multipath interference is prevented or reduced by electronic elements including at least one equalizer for processing a plurality of audio signals received along the plurality of radio paths and performing a signal correction.