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.

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 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: 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: 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).

Abstract: A system having a head-mounted display (HMD) mount and a mobile device, are described. A processor of the system can determine whether the mobile device is mounted on the HMD mount and handle an audio signal communicated from the mobile device to a wireless headphone based on whether the mobile device is mounted on the HMD mount. When the mobile device is not mounted on the HMD mount, the mobile device or the wireless headphone may operate in a first audio mode. When the mobile device is mounted on the HMD mount, the mobile device or the wireless headphone may operate in a second audio mode. The second audio mode can reduce audio signal latency between the mobile device and the wireless headphone and increase motion-to-sound quality. Other embodiments are also described and claimed.

Abstract: A presence of a person within a camera field of view of an electronic device is determined by digitally processing images captured by a camera. A position of a body member of the person with respect to the electronic device is also computed by digitally processing the camera captured images. A crosstalk cancellation (XTC) signal is adjusted based on the computed position of the body member. Adjusting the XTC signal includes adjusting a first predetermined model location, which includes a location at which a user should be in order to achieve a desired virtual acoustics effect. Processing program audio based on the adjusted XTC signal, to generate audio signals that drive speakers. Other aspects are also described and claimed.

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: An auto voice trigger method and an audio analyzer employing the same are provided. The auto voice trigger method includes: receiving a signal by at least one resonator microphone included in an array of a plurality of resonator microphones with different frequency bandwidths; analyzing the received signal and determining whether the received signal is a voice signal; and when it is determined that the received signal is the voice signal, waking up a whole system to receive and analyze a wideband signal.

Abstract: [Object] To collect a target sound in a more suitable aspect even under an environment in which noise occurs at random. [Solution] An information processing device including: a sound collection unit; and a holding member configured to have a projection portion with a streamline shape in at least a part and hold the sound collection unit so that the sound collection unit is located at a front end or near the front end of the projection portion.

Abstract: Provided is a sound pick-up device which includes: a housing having a sound hole; a main microphone which is disposed in the inside of the housing, receives sound pressure from the outside of the housing through the sound hole, and generates a first sound signal; a reference microphone which is disposed in the inside of the housing and in a proximity of the main microphone and generates a second sound signal; a first shield which blocks between the inside of the housing and the inside of the main microphone; a second shield which blocks between the outside of the housing and the inside of the reference microphone; and a third shield which blocks between the inside of the housing and the inside of the reference microphone.

Abstract: Methods and systems are provided for enhanced audio experiences in VR/AR applications. The apparatuses of this disclosure are adapted to record multiple binaural stereo pairs and play back select binaural pairs corresponding to users' head positions. A substantially spherical microarray is utilized in various embodiments for recording multiple binaural stereo pairs. A VR/AR headset is further adapted to track a user's head positions and dynamically play back binaural sound pairs corresponding to the head positions.

Abstract: A portable communication device implements a position indication method to determine and convey the relative locations of other portable communication devices in a group. In one embodiment, the portable communication devices are equipped with a GPS sensor and the position indication method determines relative locations of devices using GPS location data. In another embodiment, the portable communication devices determine relative locations of devices using the strength of the received signal as a proxy for distance. In some embodiments, the relative location information is conveyed to each portable communication device by modifying the audio signal playback. For example, a 3D audio playback may be used to convey the relative location information. In other embodiments, other non-audible location indication may be provided on each portable communication device.

Abstract: A microphone device includes a microphone array configured to capture one or more audio objects associated with a three-dimensional sound field. The microphone array includes clusters of two or more microphone elements. Each cluster includes one or more acoustic port openings and two or more microphone elements coupled to the one or more acoustic port openings via corresponding acoustic ports. The microphone device also includes a processor coupled to the microphone array.

Abstract: A hearing device, such as a hearing aid, includes an antenna for wireless communication. The antenna is housed in the hearing aid with an orientation determined to approximately minimize change in performance of the wireless communication when the hearing aid goes onto a wearer's head from free space. In various embodiments, the orientation of the antenna can be optimized by considering various factors including head loading and performance of wireless communication with various other devices.

Abstract: Detection of a blocked microphone involves receiving microphone signals from a plurality of microphones. A plurality of signal feature measures are derived from the microphone signals. The signal feature measures are normalised. The normalised signal feature measures are variably weighted in response to detected environmental conditions in the microphone signals. The variably weighted normalised signal feature measures are combined to produce an output indication of whether a microphone is blocked.

Abstract: There is a need to enable a wireless device to output signals such that the natural instincts of the user may be triggered. The present disclosure provides a solution by enabling a wireless device to determine a time delay and intensity difference associated with an audio output at a subset output devices based at least in part on a perceived spatial placement of at least one signal, a position of the transmitting device, and/or a signal strength associated with the signal. Using the time delay and the intensity difference, the output devices farthest from the transmitting device may output the at least one signal using the time delay and the intensity difference such that the user may detecting differences in the sound wave's time of arrival and intensity at the left ear versus the right ear, and thus, perceive the location of the audio source.

Abstract: A microphone with integrated speaker which includes a front casing with a front connecting portion and a front handle portion having a battery opening; a rear casing with a rear connecting portion and a rear handle portion connected to the front casing to form a first cavity with a cylindrical connecting portion and a cylindrical handle portion with a second cavity; a metallic net cover coupled with the cylindrical connecting portion; a vocal pickup support arrangement provided inside the metallic net cover between the front connecting portion and the rear connecting portion in which a vocal pickup unit is provided; a speaker positioned inside the first cavity; a pair of electrode plates positioned inside the second cavity; a circuit board affixed on the front casing; a battery cover detachably connected to the front handle portion; and an aluminum alloy handle covers the cylindrical handle portion.

Abstract: Embodiments of the disclosure generally include a method and apparatus for receiving and separating unwanted external noise from an audible input received from an audible source using an audible signal processing system that contains a plurality of audible signal sensing devices that are arranged and configured to detect an audible signal that is received from any position or angle within three dimensional space. The audible signal processing system is configured to analyze the received audible signals using a first signal processing technique that is able to separate unwanted low frequency range noise from the received audible signal and a second signal processing technique that is able to separate unwanted higher frequency range noise from the received audible signal. The audible signal processing system can then combine the signals processed by the first and second signal processing techniques to form a desired audible signal that has a high signal-to-noise ratio throughout the full speech range.

Abstract: This disclosure describes voice-enabled modular devices that act as functional, voice-controlled endpoints at which users may provide voice commands. The voice-enabled modular devices may include a power unit which may be installed into a wall permanently to serve as a power source, and a front-panel module which detachably couples to the power unit via a hardware interface. The hardware interface may allow the front-panel module and power unit to communicate with each other, and may further allow the power unit to provide power to the front-panel module. The power unit of the voice-enabled modular device described herein may be configured to provide power to multiple front-panel modules having different capabilities, and potentially different power requirements, thereby supporting multiple front-panel modules to increase the functionality of the voice-enabled modular devices.

Abstract: A system is provided for identifying a source of an audible nuisance in a vehicle. The system includes a device configured to receive a visual dataset, and to generate a camera signal in response thereto. The system includes a dock configured to removably couple to the device. The dock includes a microphone array configured to receive the audible nuisance, and to generate a microphone signal in response thereto. The system includes a processor module configured to be communicatively coupled with the device and the dock. The processor module is configured to generate a raw soundmap signal in response to the microphone signal. The processor module is configured to combine the camera signal and the raw soundmap signal, and to generate a camera/soundmap overlay signal in response to combining the camera signal and the raw soundmap signal.

Abstract: A dummy head has facial features that resemble a human face and includes a microphone inside a left ear and a microphone inside a right ear. A configuration of the facial features of the dummy head changes to enable the dummy head to capture binaural sound with different head related impulse responses (HRIRs).

Abstract: A method captures binaural sound of a voice of a first user with microphones located at left and rights ears of a dummy head. The dummy head transmits the voice of the first user to a portable electronic device with or near the first user. This portable electronic device transmits the binaural sound over one or more networks to another electronic device being used by a second user to communicate with the first user during the electronic call.

Abstract: Technology presented herein increases a user's enjoyment of sound by personalizing an audio signal so that the user perceives the audio signal as if the user had ideal hearing and/or desired hearing. In one embodiment, headphones on a user's head include a sensor and a speaker. While the speaker plays an audio signal to the user, the sensor records the user's response to the audio signal. The sensor can be a microphone, a brainwave sensor, an EEG sensor, etc. The user's response can be the audio response inside the user's ear, the brainwave response associated with the user, electrical skin response associated with the user, etc. Based on the measured response, and based on the knowledge of how other people perceive sound, the audio signal is modified to compensate for the difference between the user's hearing and the ideal hearing and/or desired hearing, therefore increasing the user's enjoyment of sound.

Abstract: An electronic device includes a vibration generation unit configured to generate vibration and a switching unit configured to perform switching between a first mode for causing the vibration generation unit to generate the vibration on the basis of first vibration information including information indicating an amplitude and a frequency, and a second mode for causing the vibration generation unit to generate the vibration on the basis of second vibration information including information indicating an amplitude and a frequency, wherein the second vibration information is data obtained by reducing an amplitude of a predetermined frequency band in the first vibration information.

Abstract: Methods and systems for generating an audio mix indicative of action sound captured at an event on a surface (e.g., a sporting event on a field) using a microphone array, where the array includes subsurface microphones (e.g., a large number of subsurface microphones) positioned under the surface, and optionally also other microphones. In typical embodiments, at least one point of interest (“PI”) on the surface is selected in an automated manner, PI data indicative of a currently selected PI on the surface is generated (e.g., a sequence of PIs on the surface is selected, the PI data is indicative of the sequence of PIs, and a most recently selected PI in the sequence is the currently selected PI), and the audio mix is generated in response to the PI data. Aspects include methods performed by any embodiment of the system, and a system or device configured (e.g., programmed) to perform any embodiment of the method.

Abstract: At a microphone array, a soundfield is detected to produce a set of microphone signals each from a corresponding microphone in the microphone array. The set of microphone signals represents the soundfield. The detected soundfield is decomposed into a set of sub-soundfield signals based on the set of microphone signals. Each sub-soundfield signal is processed, such that each sub-soundfield signal is separately dereverberated using other ones of the sub-soundfield signals to remove reverberation from the sub-soundfield signal, to produce a set of processed sub-soundfield signals. The set of processed sub-sound field signals are mixed into a mixed output signal.

Abstract: A method is disclosed for sensing, reporting, and correcting microphone proximity for a headset. The method includes determining a current proximity of a headset microphone to a face of a user. The method also includes storing the current proximity of the headset microphone as a calibrated microphone proximity value, and selecting, based on the calibrated microphone proximity value, a range of microphone proximities. Further, the method includes sensing a new proximity of the headset microphone, and comparing the new proximity of the headset microphone with the range of microphone proximities. Still yet, the method includes generating a message for the user in response to determining, based on the comparison, that the new proximity of the headset microphone is outside the range of microphone proximities.

Abstract: The disclosure relates to an audio signal processing apparatus comprising a determiner being configured to determine a filter matrix C on the basis of an acoustic transfer function matrix H and a target acoustic transfer function matrix VH, wherein the acoustic transfer function matrix H comprises transfer functions of acoustic propagation paths between loudspeakers and a listener and the target acoustic transfer function matrix VH comprises target transfer functions of target acoustic propagation paths, wherein the target acoustic propagation paths are defined by a target arrangement of virtual loudspeaker positions relative to the listener, a filter being configured to filter the input audio signal on the basis of the filter matrix C to obtain filtered input audio signals, and a combiner being configured to combine the filtered input audio signals to obtain output audio signals.

Abstract: A method captures binaural sound of a voice of a first user with microphones located at left and rights ears of a dummy head. The dummy head transmits the voice of the first user to a portable electronic device with or near the first user. This portable electronic device transmits the binaural sound over one or more networks to another electronic device being used by a second user to communicate with the first user during the electronic call.

Abstract: A dummy head has facial features that resemble a human face and includes a microphone inside a left ear and a microphone inside a right ear. A configuration of the facial features of the dummy head changes to enable the dummy head to capture binaural sound with different head related impulse responses (HRIRs).

Abstract: Measures are described by which the back volume of a microphone component can be realized regardless of its packaging. Within the framework of a microphone module, a circuit board is used for the 2nd-level mounting of at least one microphone component part, in whose surface at least one connection opening is formed, which terminates in a cavity in the layer structure of the circuit board. In addition, the circuit board surface having the connection opening is configured for a sealing mounting of the microphone component part above the connection opening, so that the microphone component is acoustically connected to the cavity in the circuit board via the connection opening in the circuit board surface, and this cavity functions as backside volume for the microphone component part.

Abstract: Provided is an audio system in which an audio signal to be heard can be automatically set to an appropriate volume or sound quality. The audio system is provided with: a mobile terminal device provided with a microphone unit that collects surrounding sound waves, and a transmission unit that, on the basis of the sound waves collected by the microphone unit, sets the volume or sound quality of an audio signal to be amplified by an audio device, and transmits the set audio signal volume or sound quality information to the audio device; and an audio device provided with an amplification unit that amplifies the audio signal, a control unit that adjusts the audio signal volume or sound quality by means of the amplifier, and a receiving unit that receives the audio signal volume or sound quality setting information from the mobile terminal device.

Abstract: Method and systems are provided for interpreting speech data. A method and system for recognizing speech involving a filter module to generate a set of processed audio data based on raw audio data; a translation module to provide a set of translation results for the raw audio data; and a decision module to select the text data that represents the raw audio data. A method for minimizing noise in audio signals received by a microphone array is also described. A method and system of automatic entry of data into one or more data fields involving receiving a processed audio data; and operating a processing module to: search in a trigger dictionary for a field identifier that corresponds to the trigger identifier; identify a data field associated with a data field identifier corresponding to the field identifier; and providing content data associated with the trigger identifier to the identified data field.

Abstract: A mobile multi-function device that includes a speaker, two or more microphones, and a beamformer processor is described. The beamformer processor uses the microphones to perform beamforming operations. One of the microphones shares a receiver acoustic opening with the speaker while the other microphone uses a separate acoustic opening. The receiver acoustic opening may be an earpiece opening that is held to the ear of a user while conducting a phone call with the device and provides acoustic input and output paths for the microphone and the speaker, respectively.

Abstract: A microphone unit for stereo recording capable of switching an arrangement of left and right microphones between a true XY technique and another technique. A microphone unit for stereo recording, including: a pair of unidirectional left and right microphones; and a retractable mechanism joined to the respective left and right microphones. The retractable mechanism is configured—to allow change in a horizontal angle and a vertical position of each of the left and right microphones.

Abstract: A microphone includes: first and second bi-directional microphone units having respective directional axes arranged on two straight lines passing through one point and radially extending with an interval of 120 degrees; a third bi-directional microphone unit having a directional axis arranged on a straight line perpendicular to a plane formed by the two straight lines; and an omnidirectional microphone unit arranged in sound collection regions of the first, second, and third bi-directional microphone units.

Abstract: There is provided a stereo microphone unit having a first and a second interference tube which are arranged at an angle relative to each other and which respectively have a first end and a second end. The stereo microphone unit has a first and a second microphone capsule for detecting an audio signal. The microphone capsules are respectively provided at an end of the first and second interference tubes. The microphone unit further has a securing unit for jointly securing the first and second interference tubes, in particular to a mobile device or to a stand. There is also a cover as a wind protector which is connected to the securing unit and completely encloses the first and second interference tubes.

Abstract: A system of this invention is directed to a speech processing system that efficiently performs noise suppression processing for a plurality of noise sources spreading in a lateral direction with respect to a speaker of interest.

Abstract: A system is provided for identifying a source of an audible nuisance in a vehicle. The system includes a device including a camera configured to receive a visual dataset, and to generate a camera signal in response thereto. The system includes a dock configured to removably couple to the device. The dock includes a microphone array configured to receive the audible nuisance, and to generate a microphone signal in response thereto. The system includes a processor module configured to be communicatively coupled with the device and the dock. The processor module is configured to generate a raw soundmap signal in response to the microphone signal. The processor module is configured to combine the camera signal and the raw soundmap signal, and to generate a camera/soundmap overlay signal in response to combining the camera signal and the raw soundmap signal.