Abstract: A speaker device with equalization (EQ) control is provided, wherein the speaker device may be a pair of earbuds/earphones or a headset having a pair of headphones, comprising: an audio receiving port for receiving audio signals from an audio source, a circuitry comprising at least a processor and a memory, the processor being configured to execute one or more firmware or software programs to perform tasks for processing the audio signals, including equalization of the audio signals according to one of multiple EQ settings stored in the memory, a user input terminal for detecting a user input to select one of the EQ settings; and a pair of speaker drivers for emitting sound corresponding to the processed audio signals.

Abstract: A signal processing apparatus includes: an acquisition unit configured to acquire a sound collection signal based on collection of sounds in a sound collection target region by a plurality of microphones; an identification unit configured to identify a position or a region corresponding to an object in the sound collection target region; and a generation unit configured to generate a plurality of acoustic signals corresponding to a plurality of divided areas obtained by dividing the sound collection target region based on the identified position or the identified region, using the acquired sound collection signal.

Abstract: The invention relates to a method for reproducing spatially distributed sounds of a multichannel audio signal comprising: receiving time-dependent input audio signals and performing a time-frequency transform; for each time-frequency tile, determining an active directional vector {right arrow over (Da)} and a reactive directional vector {right arrow over (Dr)} from time-frequency representations of different input channels for said time-frequency tile, determining positions of virtual sound sources (VS1, VS2, VS3) with respect to the reference listening point (O) from the active directional vector {right arrow over (Da)} and the reactive directional vector {right arrow over (Dr)}, and determining frequency signal values to each virtual sound sources (VS1, VS2, VS3), and distributing time-frequency signal values of said virtual sound sources to electroacoustic transducers on the basis of a comparison between the positions of the virtual sound sources in the virtual spatial configuration and the actual position

Abstract: A hearing aid comprises a sensor configured for detecting a focus of an end user on a real sound source, a microphone assembly configured for converting sounds into electrical signals, a speaker configured for converting the electrical signals into sounds, and a control subsystem configured for modifying the direction and/or distance of a greatest sensitivity of the microphone assembly based on detected focus. A virtual image generation system comprises memory storing a three-dimensional scene, a sensor configured for detecting a focus of the end user on a sound source, a speaker configured for conveying sounds to the end user, and a control subsystem configured for causing the speaker to preferentially convey a sound originating from the sound source in response to detection of the focus, and for rendering image frames of the scene, and a display subsystem configured for sequentially displaying the image frames to the end user.

Abstract: The technology described in this document can be embodied in a computer-implemented method that includes receiving information representing audio captured by a microphone array, wherein the information includes multiple datasets each representing audio signals captured in accordance with a sensitivity pattern along a corresponding direction with respect to the microphone array. The method also includes computing, using one or more processing devices for each of the multiple datasets, one or more quantities indicative of human voice activity captured from the corresponding direction, and generating, based at least on the one or more quantities computed for a plurality of the multiple datasets, a directional audio signal representing audio captured from a particular direction.

Abstract: To provide an information processing apparatus, an information processing method, and a program capable of aurally producing distance in a virtual three-dimensional space by using the space for a connection to a communication partner, and realizing more comfortable communication. An information processing apparatus including: a reception unit configured to receive data from a communication destination; and a reproduction control unit configured to perform control such that sound data of a space of the communication destination is reproduced from a sound output unit in a space of a communication source with an output value in accordance with separation distance between the communication destination and the communication source disposed in a virtual three-dimensional space, the output value being different for each sound source type.

Abstract: A multi-ringed differential microphone array includes a first number of microphones situated along a first substantial circle having a first radius, a second number of microphones situated along a second substantial circle having a second radius, and a processing device, communicatively coupled to the first number microphones and the second number of microphones, to receive a plurality of electronic signals generated by the first number of microphones and the second number of microphones, determine a differential order (N) based on the second number, and execute an N-th order minimum-norm beamformer to calculate an estimate of the sound source based on the plurality of electronic signals.

Abstract: In embodiments of the present invention, a method of modifying ambient sound received by an earpiece in accordance with an environmental characterization may have one or more of the following steps: (a) receiving the ambient sound at a microphone operably coupled to the earpiece, (b) determining, via a processor operably coupled to the microphone, the environmental characterization based on the ambient sound, (c) modifying, via the processor, the ambient sound in accordance with a plurality of parameters associated with the environmental characterization to create a modified ambient sound, (d) communicating, via a speaker operably coupled to the earpiece, the modified ambient sound, (e) receiving the ambient sound at the second microphone, (f) determining, via the processor, a location of the ambient sound from a temporal differential and an intensity differential between the reception of the ambient sound at the microphone and the reception of the ambient sound at the second microphone, and (g) determining,

Abstract: A voice control system for a police vehicle, including: a single azimuthally-sensitive directional microphone configured to receive an audio command for controlling an aspect of the police vehicle and identify a direction of origin of the audio command; and a controller including a processor and tangible memory encoding instructions which, when executed by the processor, cause the controller to: identify a command associated with the audio command received by the single azimuthally-sensitive directional microphone; and when the direction of origin allows for the command, initiate the command to control the aspect of the police vehicle.

Abstract: Vehicle communication systems and methods of operating the same. The vehicle communication system includes one or more microphones and one or more loudspeakers positioned about a vehicle. The system includes processor executable instructions for determining that the vehicle is operating in an emergency mode and, in response, activating a first microphone and a first loudspeaker for enabling communication between a first acoustic zone within the vehicle and a second acoustic zone exterior to the vehicle. The system may detect, from one of the first acoustic zone and the second acoustic zone, a first acoustic input and generate a second signal from the detected first acoustic input. The system may transmit the second signal to the first loudspeaker for producing sound waves that are audible in the other of the first acoustic zone and the second acoustic zone.

Abstract: A signal processing device for configurable voice activity detection. A plurality of inputs receive respective microphone signals. A microphone signal router configurably routes the microphone signals. At least one voice activity detection module receives a pair of microphone signals from the router, and produces a respective output indicating whether speech or noise has been detected by the voice activity detection module in the respective pair of microphone signals. A voice activity decision module receives the output of the voice activity detection module(s) and determines whether voice activity exists in the microphone signals. A spatial noise reduction module receives microphone signals from the microphone signal router, and performs adaptive beamforming based in part upon the output of the voice activity decision module, and outputs a spatial noise reduced output. The device permits simple configurability to deliver spatial noise reduction for one of a wide variety of headset form factors.

Abstract: A hearing system performs nonlinear processing of signals received from a plurality of microphones using a neural network to enhance a target signal in a noisy environment. In various embodiments, the neural network can be trained to improve a signal-to-noise ratio without causing substantial distortion of the target signal. An example of the target sound includes speech, and the neural network is used to improve speech intelligibility.

Abstract: The present technique relates to a sound field reproduction device, a sound field reproduction method, and a program that make it possible to further accurately reproduce a certain sound field. A feature amount extraction unit extracts a main sound source feature amount from a sound pickup signal obtained by picking up a sound from a main sound source. A main sound source separation unit separates the sound pickup signal obtained through the sound pickup with a microphone array that mainly picks up a sound from the main sound source into a main sound source component and an auxiliary sound source component using the main sound source feature amount. On the basis of the main sound source component and the auxiliary sound source component that have been separated, a main sound source emphasis unit generates a signal in which the main sound source components are emphasized. A drive signal for a speaker array is generated from the signal generated in this manner and supplied to the speaker array.

Abstract: A method includes performing, at a processor, signal processing operations on signals captured by each microphone in a microphone array. The method also includes performing a first directivity adjustment by applying a first set of multiplicative factors to the signals to generate a first set of ambisonic signals. The first set of multiplicative factors is determined based on a position of each microphone in the microphone array, an orientation of each microphone in the microphone array, or both.

Abstract: A single-ended signal transmission system recovers a noise signal associated with a data input signal and uses the recovered noise signal to compensate for noise on the data input signal. The noise signal may be recovered from a noise reference signal line, or clock signal line, or a data signal line associated with a DC-balanced data input signal. The recovered noise signal may be represented as an analog signal or a digital signal. The recovered noise signal may be processed to compensate for DC offset and nonlinearities associated with one or more different input buffers. In one embodiment, the recovered noise signal includes frequency content substantially below a fundamental frequency for data transmission through the data input signal.

Abstract: Presented herein are signal processing techniques that integrate an adaptive filtering process with a parametric post-filter in the frequency domain to control the amount of body noise reduction in each of a plurality of frequency bands (e.g., tuneable body noise reduction in each frequency band). The parametric post-filter generates a gain mask that is tuned to responses of sensors to separate external acoustic sounds and body noises, while limiting distortions of own voice. The techniques presented herein can preserve external acoustic sound quality, attenuate own voice to a comfortable level without distortion, suppress body noises, and/or lower a noise floor of the output signal.

Abstract: An apparatus comprising: an input configured to receive at least three microphone audio signals, the at least three microphone audio signals comprising at least two near microphone audio signals generated by at least two near microphones located near to an desired audio source and at least one farmic audio signal generated by a farmic located further from the desired audio source; a first interference canceller module configured to generate a first processed audio signal based on a first selection from the near microphone audio signals; at least one further interference canceller module configured to generate at least one further processed audio signal based on at least one further selection from the at least three microphone audio signals; a comparator configured to determine from the first processed audio signal and the at least one further processed audio signal the audio signal with greater noise suppression.

Abstract: A noise monitoring system has at least two microphones and a microphones sensitivity register. The microphones are arranged physically with respect to each other such that when the microphones are placed within an acoustic field to be monitored, the RMS signal level of that acoustic field at each of the microphones is substantially the same for a monitoring frequency range. The microphones sensitivity register has reference data indicative of an initial sensitivity difference (?0) between output RMS signal levels measured by the microphones in a reference acoustic field. A current difference (?1, ?2) between the RMS signal levels concurrently output by the microphones for the currently monitored acoustic field is compared. A fault signal is generated if the current difference (?1, ?2) differs from the initial sensitivity difference (?0) more than an acceptable limit.

Abstract: A method for estimating directions of arrival of audio signals based on categories is disclosed herein. The method includes receiving an audio signal; generating a plurality of filters based on the audio signal, each of the filters corresponding to one of a plurality of sound content categories; separating the audio signal into a plurality of content-based audio signals by applying the filters to the audio signal, each of the content-based audio signals contains a content of a corresponding sound content category among the plurality of sound content categories; for each of the content-based audio signals, generating a direction of arrival (DOA) for a sound source of the content-based audio signal.

Abstract: The invention concerns a microphone probe having a body being substantially a first solid of revolution with a number of audio sensors distributed thereon and located in the recesses. The recesses have substantially a shape of a second body of revolution with an axis of symmetry perpendicular to the surface of the body. The sensors are connected to an acquisition unit, that delivers audio signals to the output. The audio sensors are digital audio sensors comprising printed circuit board with MEMS microphone element mounted thereon, wherein MEMS microphone element is mounted on the side of the printed circuit board facing the inner side of the body, so that the sound reaches MEMS microphone element via recess and opening. The depth of recesses is in a range between 3 and 20 mm. The acquisition unit has a clocking device determining common time base for audio sensors.

Abstract: Example embodiments disclosed herein relate to assessment and adjustment for an audio environment. A computer-implemented method is provided. The method includes obtaining a first audio signal captured by a device located in an environment. The method also includes analyzing a characteristic of the first audio signal to determine an acoustic performance metric for the environment. The method further includes, in response to the acoustic performance metric being below a threshold, providing a first task for a user to perform based on the characteristic of the first audio signal. The first task is related to an adjustment to a setting of the environment. Embodiments in this regard further provide a corresponding computer program product. Corresponding system and computer program product are also disclosed.

Abstract: A mechanism is described for facilitating simultaneous recognition and processing of multiple speeches from multiple users according to one embodiment. A method of embodiments, as described herein, includes facilitating a first microphone to detect a first speech from a first speaker, and a second microphone to detect a second speech from a second speaker. The method may further include facilitating a first beam-former to receive and process the first speech, and a second beam-former to receive and process the second speech, where the first and second speeches are at least received or processed simultaneously. The method may further include communicating a first output associated with the first speech and a second output associated with the second speech to the first speaker and the second speaker, respectively, using at least one of one or more speaker devices and one or more display devices.

Abstract: The equipment of micro-acoustic detection analysis and the processing method of audio signal array based on this equipment, which is the field of micro-acoustic detection, could give a solution that having an excessive sound intensity lower limit using capacitor micro-acoustic detector, and using current way of audio signal processing could not identification and positioning for sound source from micro-acoustic detector at same time. This equipment detects sound pressure based on micro-acoustic detection cell of graphene membrane. The method includes separate the noise from audio signal, compare the extract audio feature and sound source position information with multi sound source signal stored in memory, identifying the target sound source, and the procedure of positioning the sound source according the sound source position information. This invention could be applied to the micro-acoustic detection and then identification and positioning using the detected sound source.

Abstract: A microphone equalization system determines an equalization filter which is used to spectrally shape a signal from a microphone. A loudspeaker has an enclosure, a back volume, a driver diaphragm, and a sensor located inside the back volume. An external microphone is located outside the loudspeaker. A processor determines the equalization filter based on comparison of a first signal from an output of the sensor and a second signal from an output of the external microphone both of which are produced while the driver diaphragm produces acoustic pressure waves produced by the driver diaphragm. The processor is to then spectrally shape a third signal from the output of the external microphone that is responsive to further acoustic pressure waves from a source other than the loudspeaker. Other aspects are also described and claimed.

Abstract: The present disclosure relates to a pre-5th-Generation (5G) or 5G communication system to be provided for supporting higher data rates beyond 4th-Generation (4G) communication system such as Long Term Evolution (LTE). The present disclosure provides a differential beamforming based random access method, base station, and user equipment, wherein the differential beamforming based random access method comprises, by a base station: receiving a preamble sequence from a first terminal in a differential beamforming receiving mode; determining a base station beam direction angular deviation based on the preamble sequence; and adjusting a base station beam according to the base station beam direction angular deviation, and transmitting a random access response signal to the first terminal through the adjusted base station beam.

Abstract: The present technology relates to a signal processing device, a signal processing method, and a program that can provide sound with more increase in realistic sense by restoring a difference of sound pressure between sound sources. The signal processing device includes: an acoustic signal obtaining unit that obtains an acoustic signal; a sound source positional information obtaining unit that obtains sound source positional information relating to a distance between a sound source and a sound collecting unit that collects sound from the sound source; a listening positional information obtaining unit that obtains listening positional information relating to a distance between a reproducing device that reproduces the acoustic signal and a listener who listens to sound reproduced by the reproducing device; and a correcting unit that corrects gain of the acoustic signal on a basis of the sound-source positional information and the listening positional information. The present technology can be applied to, e.g.

Abstract: The present invention makes it possible to reduce noise from a driving unit with a two-channel microphone configuration. An audio processing apparatus has a first microphone whose main acquisition target is sound from outside of the apparatus, a second microphone whose main acquisition target is driving noise from the driving unit, and a noise removing unit that generates two-channel audio data in which driving noise made by the driving unit of the apparatus has been reduced based on the difference between time-series audio data acquired by the first microphone and time-series audio data acquired by the second microphone. This noise removing unit has two adaptive filter units that respectively perform filter processing on time-series audio data from the first microphone and time-series audio data from the second microphone, and generates stereo two-channel audio data.

Abstract: Systems and methods of determining a transmitter or receiver configuration in a wireless communication system are provided. In one exemplary embodiment, a method performed by a radio node (101, 111, 200, 300a-b, 500, 700a-b) in a wireless communication system (100) comprises determining (405) a first transmitter or receiver configuration for transmission or reception in a first frequency spectrum (121) having a first sub-carrier spacing (123) based on a channel information measurement in a second frequency spectrum (131) having a second sub-carrier spacing (133), wherein the first (123) and second (133) sub-carrier spacings are different and the first (121) and second (131) frequency spectrums overlap.

Abstract: Focusing sound signals in a shared 3D space uses an array of physical microphones, preferably disposed evenly across a room to provide even sound coverage throughout the room. At least one processor coupled to the physical microphones does not form beams, but instead preferably forms 1000's of virtual microphone bubbles within the room. By determining the processing gains of the sound signals sourced at each of the bubbles, the location(s) of the sound source(s) in the room can be determined. This system provides not only sound improvement by focusing on the sound source(s), but with the advantage that a desired sound source can be focused on more effectively (rather than steered to) while un-focusing undesired sound sources (like reverb and noise) instead of rejecting out of beam signals. This provides a full three dimensional location and a more natural presentation of each sound within the room.

Abstract: An apparatus, method, and system each of which obtains sound data based on a plurality of sound signals respectively output from a plurality of microphones, receives a user instruction for enhancing directivity of sensitivity characteristics of at least one of the plurality of microphones in a specific direction, and generates sound data having the directivity in the specific direction, based on the obtained sound data.

Abstract: Apparatus including a processor configured to: receive a spatial audio signal associated with a microphone array configured to provide spatial audio capture and at least one additional audio signal associated with an additional microphone, the at least one additional microphone signal having been delayed by a variable delay determined such that the audio signals are time aligned; receive a relative position between a first position associated with the microphone array and a second position associated with the additional microphone; generate at least two output audio channel signals by processing and mixing the spatial audio signal and the at least one additional audio signal based on the relative position between the first position and the second position such that the at least two output audio channel signals present an augmented audio scene.

Abstract: An electronic device comprising: a microphone array including at least three microphones; and at least one processor configured to: identify a kind of an application that is executed; activate one or more of the microphones in the array based on each microphone's respective position within the electronic device and the type of the application; and capture audio using the activated microphones.

Abstract: A sound reproducing method used in sound reproducing apparatus that includes the steps outlined below is provided. A sound signal with a three-dimensional (3D) sound generating process is generated according to listener data and sound data. Whether a sound source position is within a target region relative to a listener position within a virtual environment is determined according to the listener data and the sound data. The sound signal is multiplied by an adjusting function to enhance peaks and valleys of the sound signal while maintaining a behavior of the sound signal when the sound source position is within the target region. The sound signal is reproduced.

Abstract: The invention relates to a marine surface vessel (1), in particular a passenger ship, extending in a longitudinal direction between a bow (2) and a stern (3), comprising —least one power providing apparatus (6) operatively connected to at least one propulsion unit (7) for propulsion of the vessel, —a fuel tank arrangement (8) for storing gaseous fuel in a liquid phase, connected to the power providing apparatus (6) by means of a gas supply system for running the power providing apparatus (6), and —a plurality of decks (91-98) distributed vertically, one above the other, —the fuel tank arrangement (8) comprising two tank sub-arrangements (81, 82), each tank sub-arrangement comprising at least one gaseous fuel tank (811, 812, 821, 822), each fuel tank extending vertically past at least two of the decks (92-95), and at least two of the decks (92-95), past which the fuel tanks (811, 812, 821, 822) extend, each presents a passageway (941) extending between the tank sub-arrangements (81, 82).

Abstract: The disclosed technology relates to a microphone array. The array comprises a plurality of microphones with each microphone having a horn portion. Each microphone of the array further comprises an instrument disposed at a distal end of the horn portion. Each instrument of the array is configured to convert sound waves into an electrical signal. The microphone array further comprises a beamforming signal processing circuit electrically coupled to each instrument and configured to create a plurality of beam signals based on respective electrical signals.

Abstract: Embodiments disclosed herein provide systems and methods for pairwise audio capture device selection. In a particular embodiment, a method provides capturing audio information via a plurality of audio capture device pairs, wherein each audio capture device pair comprises a first audio capture device and a second audio capture device. Based on the audio information, the method provides determining a difference in audio energy between audio captured by the first and second audio capture devices for each of the plurality of audio capture device pairs. The method further provides selecting an audio capture device pair of the plurality of audio capture device pairs based on the audio capture device pair having the largest difference in audio energy.

Abstract: A method for providing sound detection information includes steps of: sensing sound around a host vehicle to generate sound data; generating a result of sound detection based on the sound data; calculating a rate of change based on the result of sound detection; generating a result of tunnel detection by comparing the rate of change with a threshold; and controlling at least one peripheral apparatus in the host vehicle according to the result of tunnel detection.

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: 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: A sound information display device including: a plurality of sound collecting units; an information obtaining unit that obtains positional information and sound information of a sound source based on collected sound information of the plurality of sound collecting units; a display data generation unit that generates display data for displaying sound information that is obtained by the information obtaining unit overlaid on a visual image at a position that corresponds to positional information that is obtained by the information obtaining unit within the visual image; and a head-mounted image display unit that displays sound information of a sound source overlaid on the visual information at a position that corresponds to the sound source within the visual image based on display data that is generated by the display data generation unit.

Abstract: There is provided a post processing technique or method for separation algorithms to separate vocals from monaural music recordings. The method comprises detecting traces of pitched instruments in a magnitude spectrum of a separated voice using Hough transform and removing the detected traces of pitched instruments using median filtering to improve the quality of the separated voice and to form a new separated music signal. The method further comprises applying adaptive median filtering techniques to remove the identified Hough regions from the vocal spectrogram producing separated pitched instruments harmonics and new vocals while adding the separated pitched instruments harmonics to a music signal separated using any separation algorithm to form the new separated music signal.

Abstract: A method includes: receiving, by using channels located in different positions of a terminal device, at least three signals emit by a same sound source; determining, according to three signals in the at least three signals, a signal delay difference between every two of the three signals; determining, according to the signal delay difference, the position of the sound source relative to the terminal device; and when the sound source is located in front of the terminal device, performing orientation enhancement processing on a target signal in the at least three signals, and obtaining a first output signal and a second output signal of the terminal device according to a result of the orientation enhancement processing, where the orientation enhancement processing is used to increase a degree of discrimination between a front characteristic frequency band and a rear characteristic frequency band of the target signal.

Abstract: A sound collecting apparatus capable of effectively suppressing sounds other than a target sound is provided. The sound collecting apparatus includes a plurality of microphones. A total number of effective microphone pairs in which a distance between two microphones is smaller than a distance D is larger than a total number of the plurality of microphones. The distance D is represented by D=c/2f, where the frequency of the target sound acquired from each of the plurality of microphones is f and sound velocity is c. When an angle formed by a straight line connecting two microphones configuring an effective microphone pair and a predetermined straight line is ?, the angles ? of all effective microphone pairs acquired from the plurality of microphones are different from each other.

Abstract: In certain embodiments, an article of manufacture, such as a cell phone, has a device body with a non-spheroidal shape, such as a parallelepiped, and microphones configured at different locations on the device body. A signal processing system processes the microphone signals to generate a plurality of different output beampatterns in at least two non-parallel directions, wherein, in generating at least one of the output beampatterns, the signal processing system takes into account effects of the device body on the incoming acoustic signal. Four or more microphones can be used to generate B format output beampatterns, such as three dipole beampatterns and an omnidirectional beampattern.

Abstract: A method determines a direction of a useful signal source in an acoustic system that has a first input transducer and a second input transducer. The first input transducer generates a first input signal from a sound signal from the surroundings and the second input transducer generates a second input signal from the sound signal. The first input signal and the second input signal are used to form a plurality of angle-dependent directional characteristics having a respective fixed central angle and a respective given angular expansion. The signal components pertaining to the individual directional characteristics are examined for the presence of a useful signal from a useful signal source. The applicable central angle is assigned to a useful signal source ascertained in a particular directional characteristic as the direction of the useful signal source.

Abstract: Speech enhancers suppress impairments in an acoustic signal. An audio appliance has a first microphone and a second microphone. The first microphone provides a first signal, and the second microphone provides a second signal. A voice-activity detector can determine a presence of user speech responsive to a combination of voice-activity cues, including a first level difference between the first signal and the second signal within a first frequency band, and a second level difference between the first signal and the second signal within a second frequency band. A noise suppressor suppresses impairments originating from a direction of, e.g., up to about 75-degrees from an axis extending from the second microphone to the first microphone. An output device can output a noise-suppressed output-signal corresponding to a determined presence or absence of speech by the voice-activity detector. The impairments can be suppressed by, e.g., between about 3 dB and about 20 dB.

Abstract: An apparatus and a method for measuring one or more acoustic parameters in a duct are described. The method includes arranging at least one pair of microphones in an array on a microphone support device, disposing the microphone support device including the array of microphones in the duct, generating an acoustic signal in the duct such that the acoustic signal propagates toward the array of microphones, and receiving by the array of microphones, the acoustic signal to provide a microphone output signal. The microphone support device is configured to fit within an inner diameter of the duct.

Abstract: Impulse responses of a device are measured. A database of sound files is generated by convolving source signals with the impulse responses of the device. The sound files from the database are transformed into time-frequency domain. One or more sub-band directional features is estimated at each sub-band of the time-frequency domain. A deep neural network (DNN) is trained for each sub-band based on the estimated one or more sub-band directional features and a target directional feature.

Abstract: A speech enhancement system for a remote microphone has a wireless receiver that receives a signal from a first microphone of a remote device. A delay buffer receives a second microphone signal from a second microphone and delays by an adjustable delay. The adjustable delay is based on a difference between a wireless delay and an acoustic delay. A noise suppressor produces an output audio signal for an earpiece speaker, based on the first microphone signal and the adjustable delayed second microphone signal. Other aspects are also described and claimed.

Abstract: An electronic device includes a first microphone that receives a sound generated for a specific time period, from the outside, a second microphone, which is disposed at a location spaced apart from the first microphone and which receives the sound, an audio converter comprising audio converting circuitry, and a processor electrically connected with the first microphone, the second microphone, and the audio converter. The processor is configured to convert the sound obtained from the first microphone, into a first signal and to convert the sound obtained from the second microphone, into a second signal, using the audio converter, and to determine the sound, which is generated for the specific time period, as a voice or a noise based on a frequency-related correlation between the first signal and the second signal.