Abstract: An image capture device may include a sensor, a microphone array, and a processor. The microphone array may include a first microphone, a second microphone, a third microphone, or any combination thereof. The first microphone may be configured to face a first direction. The second microphone may be configured to face a second direction. The second direction may be diametrically opposed to the first direction. The third microphone may be configured to face a third direction. The third direction may be substantially perpendicular to the first direction, the second direction, or both. The processor may be configured to determine a microphone capture pattern. The microphone capture pattern may be determined based on data obtained from the sensor. The sensor data may include image data, audio data, image capture device orientation data, location data, accelerometer data, or any combination thereof.

Abstract: A vehicle passenger locator and identifier system and method is provided herein. Once a vehicle is within a certain vicinity of an intended passenger via conventional mobile device geolocation or geofencing, and/or via a conventional camera-based technology, a plurality of microphones listen for the passenger to utter a predetermined word or phrase. Optionally, this predetermined word or phrase has been uttered to a mobile application used to summon the vehicle such that an accurate voice comparison can be made by the vehicle. Alternatively, the same or a different mobile device can be used to emit an auditory beacon that is detected by the vehicle. Once this word, phrase, or auditory beacon is detected, the passenger can be triangulated, located, identified, and even authorized for pickup. All of this is carried out via an onboard vehicle processor and software and/or a processor and software residing in the cloud.

Abstract: The description relates to rendering directional sound. One implementation includes receiving directional impulse responses corresponding to a scene. The directional impulse responses can correspond to multiple sound source locations and a listener location in the scene. The implementation can also include encoding the directional impulse responses to obtain encoded departure direction parameters for individual sound source locations. The implementation can also include outputting the encoded departure direction parameters, the encoded departure direction parameters providing sound departure directions from the individual sound source locations for rendering of sound.

Abstract: A headphone, headphone system, and speech enhancing method is provided to enhance speech pick-up from the user of a headphone and includes receiving a plurality of signals from a set of microphones and generating a primary signal by array processing the microphone signals to steer a beam toward the user's mouth. A noise reference signal is also derived from one or more microphones via a delay-and-sum technique, and a voice estimate signal is generated by filtering the primary signal to remove components that are correlated to the noise reference signal.

Abstract: A sound collecting apparatus includes a base of a substantially spherical body and a plurality of microphones provided on the base, a number of the microphones having a predetermined constraint, the plurality of microphones being alternately arranged vertically relative to a horizontal plane including a center of the substantially spherical body in order to improve resolution in a horizontal direction.

Abstract: A microphone array system includes first microphones disposed along a first axis, second microphones disposed at equal intervals of a first distance from the first axis along a second axis orthogonal to the first axis, a beamforming processor that performs beamforming by filtering and combining audio signals from microphones, and, when the second microphones are projected onto the first axis, the first microphones and projected second microphones are disposed at equal intervals of a second distance, a distance between two microphones disposed at opposite ends, among the first microphones and the projected second microphones arranged along the first axis when the second microphones are projected onto the first axis, is larger than a distance between two microphones disposed at opposite ends, among the first microphones and the projected second microphones arranged along the second axis when the first microphones are projected onto the second axis.

Abstract: An active road noise control system method for a vehicle includes picking up noise at a multiplicity of positions in or on the vehicle and generating a multiplicity of noise sense signals representative of road noise originating from a road noise source in or at the vehicle, and processing, according to a beamforming scheme, the multiplicity of noise sense signals to generate a reference signal and to provide a sensitivity characteristic for picking up the noise that comprises one main lobe directed to the road noise source. The system and method further includes iteratively and adaptively processing the reference signal to provide a noise reducing signal, and generating at one or more positions in an interior of the vehicle, from the noise reducing signal, noise reducing sound at a listening position in the interior of the vehicle.

Abstract: A video conferencing device for video conferencing between at least one local participant and a remote participant includes a video camera, a microphone array, and a speaker tracker. The video camera provides a local video input signal. The microphone array provides a local audio input signal. The speaker tracker is configured to identify a local speaker from the at least one local participant using a sound source localizer. The video conferencing device processes the local video input signal without the local speaker, based on the video conferencing device receiving a signal from a computing system, the signal dependent on a loopback audio output signal indicating that the remote participant is speaking.

Abstract: An apparatus, electronic device, method and computer program wherein the apparatus includes: processing circuitry; and memory circuitry including computer program code, the memory circuitry and the computer program code configured to, with the processing circuitry, enable the apparatus to perform: obtaining spatial information relating to a captured sound field from a first set of microphones; obtaining one or more signals from a second set of microphones where the one or more signals relate to the captured sound field; and using the obtained spatial information from the first set of microphones to process the one or more signals obtained from the second set of microphones; wherein the first set of microphones is provided within an electronic device and the second set of microphones is provided external to the electronic device.

Abstract: Techniques for speech processing using a deep neural network (DNN) based acoustic model front-end are described. A new modeling approach directly models multi-channel audio data received from a microphone array using a first model (e.g., multi-geometry/multi-channel DNN) that is trained using a plurality of microphone array geometries. Thus, the first model may receive a variable number of microphone channels, generate multiple outputs using multiple microphone array geometries, and select the best output as a first feature vector that may be used similarly to beamformed features generated by an acoustic beamformer. A second model (e.g., feature extraction DNN) processes the first feature vector and transforms it to a second feature vector having a lower dimensional representation. A third model (e.g., classification DNN) processes the second feature vector to perform acoustic unit classification and generate text data. The DNN front-end enables improved performance despite a reduction in microphones.

Abstract: Proposed is a method for determining, by a movable robot, a position of a speaker, wherein the movable robot includes first to fourth microphones installed at four vertexes of a quadrangle of a horizontal cross section of the robot respectively, wherein the method includes: receiving a wake-up voice through first and third microphones disposed respectively at first and third vertices in a diagonal direction; obtaining a first reference value of the first microphone and a second reference value of the third microphone based on the received wake-up voice; comparing the obtained first and second reference values to select the first microphone; selecting a second microphone disposed at a second vertex, wherein the first and second microphones are on a front side of the quadrangle; calculating a sound source localization (SSL) value based on the selected first and second microphones; and tracking a position of the speaker based on the SSL value.

Abstract: A speech-based audio device may be configured to detect a user-uttered wake expression. For example, the audio device may generate a parameter indicating whether output audio is currently being produced by an audio speaker, whether the output audio contains speech, whether the output audio contains a predefined expression, loudness of the output audio, loudness of input audio, and/or an echo characteristic. Based on the parameter, the audio device may determine whether an occurrence of the predefined expression in the input audio is a result of an utterance of the predefined expression by a user.

Abstract: A binaural hearing system comprises a first and second hearing aids. The hearing aids each comprises antenna and transceiver circuitry allowing the exchange of audio signals between them. At least one 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. In an embodiment, the spatially filtered signal of the secondary 2-channel beamformer may comprise an estimate of user's own voice. In an embodiment, the spatially filtered signal of the secondary 2-channel beamformer may comprise an estimate of a target signal in the environment. In an embodiment, the inputs to the secondary 2-channel beamformer may be beamformed signals from the first and second hearing aids respectively.

Abstract: A method for off-loading tasks between a set of wireless earpieces in an embodiment of the present invention may have one or more of the following steps: (a) monitoring battery levels of the set of wireless earpieces, (b) determining the first wireless earpiece battery level and the second wireless battery level, (c) communicating the battery levels of each wireless earpiece to the other wireless earpiece of the set of wireless earpieces, (d) assigning a first task involving one or more of the following: computing tasks, background tasks, audio processing tasks, and sensor data analysis tasks from one of the set of wireless earpieces to the other wireless earpiece if the battery level of the one of the set of wireless earpieces falls below a critical threshold, (e) communicating data for use in performing a second task to the other wireless earpiece if the second task is communicated to the first wireless earpiece.

Abstract: An embodiment of the present disclosure provides a method for driving an acoustic transducer, including: obtaining a reference electrical signal according to a first electrical signal output by a first acoustic transducer element in a case where sound waves are not received by the first acoustic transducer element; obtaining an actual detected electrical signal according to a second electrical signal output by a second acoustic transducer element in a case where sound waves are received by the second acoustic transducer element; and performing a noise reduction process on the actual detected electrical signal according to the reference electrical signal to obtain a noise-reduced signal as a final output electrical signal of the second acoustic transducer element in a case where sound waves are received by the second acoustic transducer element. An embodiment of the present disclosure further provides an acoustic transducer.

Abstract: A method, apparatus and computer program product provide an improved filter calibration procedure to reliably equalize the long term spectrum of the audio signals captured by first and second microphones that are at different locations relative to a sound source and/or are of different types. In the context of a method, the signals captured by the first and second microphones are analyzed. The method also determines one or more quality measures based on the analysis. In an instance in which one or more quality measure satisfy a predefined condition, the method determines a frequency response of the signals captured by the first and second microphones. The method also determines a difference between the frequency response of the signals captured by the first and second microphones and processes the signals captured by the first microphone for filtering relative to the signals captured by the second microphone based upon the difference.

Abstract: An eyewear device includes an audio system. In one embodiment, the audio system includes a microphone array that includes a plurality of acoustic sensors. Each acoustic sensor is configured to detect sounds within a local area surrounding the microphone array. For a plurality of the detected sounds, the audio system performs a direction of arrival (DoA) estimation. Based on parameters of the detected sound and/or the DoA estimation, the audio system may then generate or update one or more acoustic transfer functions unique to a user. The audio system may use the one or more acoustic transfer functions to generate audio content for the user.

Abstract: A wind noise reduction system includes a delay and sum (DAS) beamformer, an MVDR beamformer, a wind detector, a GEV beamformer, and a fixed voice mixer. The DAS beamformer generates a first voice signal based on a first and second microphone signal. The MVDR beamformer generates a second voice signal based on the first and second microphone signals. The GEV beamformer generates a wind array voice signal based on the first and second microphone signals and an accelerometer signal. The wind detector generates a wind detection signal based on the first voice signal and the second voice signal. The fixed voice mixer generates an output voice signal based on a microphone array voice signal, the wind array voice signal, and the wind detector signal. If high winds are detected, the output voice signal includes elements of the wind array voice signal based in part on the accelerometer signal.

Abstract: A method for modifying a sound produced by a sound source in a video includes capturing video and audio of a scene is disclosed. Audio is captured using a microphone array. A sound source is isolated and a direction of arrival of the sound source with respect to a capture location is identified. One or more visual objects in the captured video are identified. One of the isolated sound sources is associated with one of the identified visual objects. An input identifying one of the isolated sound sources is received during playing of the captured video and audio. The input includes a command. Responsive to receiving the input, an attribute of the identified isolated sound source is modified. The input may identify a visual object associated with a sound source. A system and article of manufacture are also disclosed.

Abstract: Disclosed are a control method and a control device for an ultrasonic receiving device. The control method includes: determining a target receiver of the ultrasonic receiving device, where the ultrasonic receiving device includes at least two ultrasonic receivers, and the target receiver is one of the at least two ultrasonic receivers on the ultrasonic receiving device that is the nearest to an ultrasonic transmitting device; and controlling a state of each of the at least two ultrasonic receivers on the ultrasonic receiving device based on the determined target receiver. Thus, the ultrasonic ranging error is reduced, and the accuracy of measurement is improved.

Abstract: An acoustic sensor assembly that produces an electrical signal representative of an acoustic signal, includes an acoustic transduction element disposed in a housing and acoustically, a heat source causing air pressure variations within the housing when energized, and an electrical circuit electrically coupled to the acoustic transduction element and to contacts on an external-device interface of the housing, wherein the electrical circuit is configured to energize the heat source and determine a non-acoustic condition or change therein based on an amplitude of air pressure variations detected by the acoustic transduction element.

Abstract: Methods and systems are provided for recording mixed audio signal and reproducing directional audio. A method includes receiving a mixed audio signal via plurality of microphones; determining an audio parameter associated with the mixed audio signal received at each of the plurality of microphones; determining active audio sources and a number of the active audio sources from the mixed audio signal; determining direction and positional information of each of the active audio source; dynamically selecting a set of microphones from the plurality of microphones based on at least one of the number of the active audio sources, the direction of each of the active audio sources, the positional information of each of the active audio sources, the audio parameter, or a predefined condition; and recording, based on the selected set of microphones, the mixed audio signal for reproducing directional audio.

Abstract: A modular ACD system is configured to automate clinical documentation and includes a machine vision system configured to obtain machine vision encounter information concerning a patient encounter. An audio recording system is configured to obtain audio encounter information concerning the patient encounter. A compute system is configured to receive the machine vision encounter information and the audio encounter information.

Abstract: Techniques for improving beamforming using filter coefficient values corresponding to virtual microphones are described. A system may define “virtual” microphone positions and determine corresponding filter coefficient values. These filter coefficient values may be applied to input audio data captured by actual physical microphones, enabling the system to improve performance of beamforming and/or to reduce a number of physical microphones without degrading performance. Offline testing and simulations may be performed to identify the best combination of virtual microphones and/or filter coefficient values for a particular look-direction. For example, the simulations may identify that a first filter coefficient corresponding to a first virtual microphone and a first direction will be associated with a first physical microphone and the first direction.

Abstract: A system that performs early reflections filtering to suppress early reflections and improve sound source localization (SSL). During music playback and/or when a device is placed in a corner, acoustic reflections from nearby surfaces get boosted due to constructive interference, negatively impacting SSL and other processing of the device. To suppress these early reflections, the device uses an Early Reflections Filter (ERF) that makes use of Linear Prediction Coding (LPC), which is already being performed during speech processing. For example, the device generates raw audio signals using multi-channel LPC coefficients and then uses single-channel LPC coefficients for each raw audio signal in order to generate a filter that estimates the reflections. The device then uses this filter to suppress the early reflections and generate filtered audio signals, thus resulting in better audio processing and better overall device performance.

Abstract: An improved method for locating an acoustic source relative to a vehicle that requires, for example, only a single microphone is disclosed. The method comprises: obtaining an acoustic signal transmitted by the acoustic source; determining an observer frequency, referenced to the vehicle, of the acoustic signal; stipulating a velocity of the acoustic source; stipulating a relative position of the acoustic source relative to a position of the vehicle; determining a signal frequency; and locating the acoustic source by performing, n times, a Doppler calculation using the determined observer frequency, the stipulated velocity, the determined signal frequency, and the stipulated relative position.

Abstract: A method of obtaining a directional microphone signal, the method comprising: receiving first and second microphone signals from first and second microphones separated by a distance; obtaining a combined microphone signal based on one or more of the first and second microphone signals; obtaining a difference microphone signal by subtracting the second microphone signal from the first microphone signal; obtaining a transformed combined microphone signal by applying a Hilbert transform to the combined microphone signal; combining the transformed combined microphone signal with the difference microphone signal to obtain the directional microphone signal.

Abstract: An apparatus, the apparatus comprising means configured to: receive audio content comprising voice audio and ambient audio and directional information indicative of a direction of the at least one sound source and the direction of the remote user relative to the reference point; receive a reference location; provide for presentation of the ambient audio with a first spatial audio effect, based on the directional information, and presentation of the voice audio with a second spatial audio effect, based on the directional information, receive repositioning signalling from the remote user device; and provide for presentation of the audio content using a modification of the first spatial audio effect to reposition an ambient-perceived direction based on the repositioning signalling and/or a modification of the second spatial audio effect to reposition a voice-perceived direction based on the repositioning signalling to increase the spatial separation between the voice-perceived direction and the ambient-perceived

Abstract: The disclosed technology generally relates to a microphone device configured to receive sound from different beams, where each beam has a different spatial orientation and is configured to receive sound from different directions. The microphone device is also configured to process the received sound using a generic or specific head related transfer function (HRTF) to generate processed audio and transmit the processed audio to hearing devices worn by a hearing-impaired user. Additionally, the microphone device can use a reference line and/or reference point when processing the received audio.

Abstract: Methods and systems for identifying human activity in a building. An illustrative method includes storing one or more room sound profiles for a room in a building based at least in part on background audio captured in the room without a presence of humans in the room. Background noise filters are generated for the room based on the room sound profiles. Real time audio may be captured from the room and filtered with at least one of the background noise filters. The filtered real time audio may be analyzed to identify one or more sounds associated with human activity in the room. A situation report may be generated based at least in part on the identified one or more sounds associated with human activity in the room and transmitted for use by a user.

Abstract: Systems, apparatuses, and methods are described for controlling source tracking and delaying beamforming in a microphone array system. A source tracker may continuously determine a direction of an audio source. A source tracker controller may pause the source tracking of the source tracker if a user may continue to speak to the system. The source tracker controller may resume the source tracking of the source tracker if the user may cease to speak to the system, or when one or more pause durations have been reached.

Abstract: Various implementations include systems for processing audio signals. In particular implementations, a system includes at least one microphone configured to capture acoustic signals; a wearable hearing assist device configured to amplify captured acoustic signals from the at least one microphone and output amplified audio signals to a transducer; a voice activity detector (VAD) configured to detect voice signals of a user from the captured acoustic signals; and a voice suppression system configured to suppress the voice signals of the user from the amplified audio signals being output to the transducer.

Abstract: A computer device including an integrated input device is provided. The integrated input device includes a positive disconnect system including a display of an open connection to the integrated input device when the integrated input device is deactivated. The open connection comprises a visible metal trace on a line coupling the integrated input device to the computer device.

Abstract: A device capable of motion includes a beamformer for determining audio data corresponding to one or more directions. The beamformer includes a target beamformer that boosts audio from a target direction and a null beamformer that suppresses audio from that direction. When the device outputs sound while moving, the target and null beamformers capture and compensate for Doppler effects in output audio that reflects from nearby surfaces back to the device.

Abstract: For certain application cases, such as e.g., in a sports stadium, a microphone array having a particularly high directivity in the vertical direction and a high, yet in wide limits adjustable directivity in horizontal direction is provided. The microphone array has a plurality of microphones whose output signals are combined into at least one common output signal. The microphones are directional microphones with a preferred direction of high sensitivity and arranged substantially in one plane on a circle or segment of a circle, such that each microphone has a different direction of high directivity. For each of the microphones, the preferred direction of high sensitivity is substantially orthogonal to the circle or segment of the circle. A common output signal of the microphone array is obtained by beamforming. The microphone array has an adjustable preferred direction of high sensitivity, wherein the common output signal comprises the sound recorded from this adjustable direction.

Abstract: In one embodiment, a video conference endpoint may detect a one or more participants within a field of view of a camera of the video conference endpoint. The video conference endpoint may determine one or more alternative framings of an output of the camera of the video conference endpoint based on the detected one or more participants. The video conference endpoint may send the output of the camera of the video conference endpoint to one or more far-end video conference endpoints participating in a video conference with the video conference endpoint. The video conference endpoint may send data descriptive of the one or more alternative framings of the output of the camera to the far-end video conference endpoints. The far-end video conference endpoints may utilize the data to display one of the one or more alternative framings.

Abstract: The present invention relates to a small spatial sound source orientation detecting device, including a circuit board; three or more MEMS acoustic-electric transducers fixedly arranged on the circuit board and centrosymmetric distribution. The distance between adjacent MEMS acoustic-electric transducers is not more than one-half of the shortest wavelength of the sound source signal; and a micro-control unit, each MEMS acoustic-electric transducer is electrically connected to the micro-control unit respectively. The micro-control unit is to obtain the orientation information of the spatial sound source based on the acoustic signals collected by three or more MEMS acoustic-electric transducers. The present invention also provides a spatial sound source orientation detection method. The small spatial sound source orientation detecting device of the present invention has an ultra-small space size, and can accurately detect the orientation information of the sound source.

Abstract: Systems and methods are provided for circuit configurations that maintain audio playback performance while reducing power consumption. In particular, a gain for a current analog-to-digital converter in an audio playback path is adjusted based on an amplitude of the input signal. Additionally, systems and methods are provided for transitioning between a modes of operation for large signals and mode of operation for small signals.

Abstract: According to one embodiment, a signal processing apparatus includes the following units. The transform unit transforms a first signal into a time-frequency domain to obtain a second signal, the first signal obtained by detecting sound at each of different positions. The first calculation unit calculates a first spatial correlation matrix based on the second signal. The second calculation unit calculates a second spatial correlation matrix based on a third signal obtained by delaying the second signal by a predetermined time. The spatial filter unit generates a spatial filter based on the first spatial correlation matrix and the second spatial correlation matrix, and filters the second signal by using the spatial filter.

Abstract: A photographing control method includes detecting a target sound from a target using a plurality of sound detectors of a mobile platform carrying a sensor, determining a target location of the target according to the target sound detected using the plurality of sound detectors, and controlling the mobile platform based at least in part on the target location.

Abstract: Some embodiments of the present disclosure provide a method and an apparatus for forming a differential beam, a method and an apparatus for processing a signal, and a chip. The method for forming a differential beam includes: obtaining a differential beam forming signal according to an input signal acquired by two microphones in a microphone array (101); and performing a nonlinear adjustment on at least an amplitude of the differential beam forming signal based on a distance between the two microphones and a signal frequency of the input signal to obtain the adjusted differential beam forming signal (102). With the above solution, a constant beam characteristic of the differential beam forming signal can be ensured as much as possible for microphone arrays of different specifications.

Abstract: A modular speaker system, comprising an exoskeleton, configured to mechanically support and quick attach and release at least one functional panel and an electrical interface provided within the exoskeleton, configured to mate with a corresponding electrical connector of the functional panel. An optional endoskeleton is provided to support internal components. The system preferably provides a digital electronic controller, and the electrical interface is a digital data and power bus, with multiplexed communications between the elements of the system. The elements of the system preferably include at least one speaker, and other audiovisual and communications components. Multiple modules may be interconnected, communicating through the electrical interface. A base module may be provided to provide power and typical control, user and audiovisual interface connectors.

Abstract: Techniques are provided for spatial audio streaming using adaptive bitrate encoding and decoding in bandwidth limited virtual reality applications. A methodology implementing the techniques according to an embodiment includes calculating weight factors for audio channels associated with each of a plurality of microphones. The weight factors are based on the position and head angle, relative to the microphones, of a user wearing a head mounted display (HMD). The method also includes encoding and decoding the audio channels, for transmission to the HMD, at a selected bitrate, the bitrate based on the weight factor for the audio channel. The method further includes applying an adaptive gain compensation to each of the received and decoded audio channels, the gain compensation also based on the weight factor. The method further includes mixing the resulting gain compensated channels to generate a stereo audio signal to be played through speakers of the HMD.

Abstract: An exemplary hearing device configured to be worn by a user includes a microphone and a processor. The microphone detects an audio signal. The processor is configured to determine that the audio signal includes own voice content representative of a voice of the user and determine that an environmental noise level within the audio signal is below a threshold The processor is further configured to apply, based on the environmental noise level being below the threshold, a biomarker feature analysis heuristic to the own voice content.

Abstract: A method for voice recognition via an earphone is disclosed. The method includes receiving first audio data via the first microphone and buffering the first audio data in response to the first trigger signal; receiving second audio data via the first microphone and recognizing whether the first audio data contains data of a wake-on-voice word in response to the second trigger signal; and recognizing whether the second audio data contains data of the wake-on-voice word. The first audio data is received and buffered in a first duration starting from when the first trigger signal is received and ending when the second trigger signal is received. The second audio data is received in a second duration starting from when the second trigger signal is received and ending when whether the first audio data contains data of the wake-on-voice word is recognized.

Abstract: An apparatus for spatial audio signal processing, the apparatus including at least one processor and at least one memory including a computer program code, the at least one memory and the computer program code configured to, with the at least one processor, cause the apparatus at least to: receive audio signals from a microphone array, the microphone array including three or more microphones forming a geometry with defined displacements between pairs of the three or more microphones; determine delay information between audio signals associated with the pairs of the three or more microphones; determine an operator based on the geometry with defined displacements between the pairs of the three or more microphones; apply the operator to the delay information to generate at least one direction parameter associated with the audio signals.

Abstract: Included are a first obtainer which obtains a first signal output from a first microphone, a second obtainer which obtains a second signal output from a second microphone installed in a position different from a position where first microphone is installed, a delayer which delays the second signal, a mixed sounds estimator which estimates noises mixed in the first signal on the basis of the second signal delayed by the delayer, and an eraser which erases the noises from the first signal, the noises being estimated by the mixed sounds estimator.

Abstract: A sound source separation system includes: a controller that: acquires pieces of sound collection data with microphones that collect sounds output from first and second sound sources. The first sound source is at a first position at which effective distances from the microphones are equal and the second sound source is at a different position. The controller further acquires, based on the sound collection data, frequency spectra in two dimensions of a circumferential direction of a circle and a time direction. The first position is a center of the circle and each of the effective distances is a radius of the circle. The controller further separates, from the frequency spectra, a first sound source spectrum and a second sound source spectrum.

Abstract: The disclosure is generally directed to a system for reducing wind noise. A system includes one or more processors coupled to a non-transitory computer-readable storage medium having instructions encoded thereon that, when executed by the one or more processors, cause the one or more processors to obtain signals respectively generated from two or more microphones during a time period, the signals representing acoustic energy detected by the two or more microphones during the time period, determine a coherence between the signals, and determine a filter based on the coherence. The filter is configured to reduce wind noise in one or more of the signals.

Abstract: An audio processing technique including obtaining a multi-channel audio signal, a first sound direction of interest and a second sound direction of interest; determining, for one or more frequency bands, a respective first range of sound directions encompassed by a first focus pattern directed to said first sound direction of interest and a respective second range of sound directions encompassed by a second focus pattern directed to said second sound direction of interest; determining, for said one or more frequency bands, respective overlap between the first and second ranges of sound directions; and deriving, based on the multi-channel audio signal and in accordance with said first and second focus patterns, a processed audio signal where sounds in said first and second sound directions of interest are emphasized in relation to sounds in other sound directions, said derivation including controlling emphasis in dependence of the determined overlap.