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

Abstract: A microphone system of the invention is applicable to an electronic device comprising an adjustable mechanism that causes a change in geometry of a microphone array. The microphone system comprises the microphone array, a sensor and a beamformer. The microphone array comprises multiple microphones that detect sound and generate multiple audio signals. The sensor detects a mechanism variation of the electronic device to generate a sensing output. The beamformer is configured to perform a set of operations comprising: performing a spatial filtering operation over the multiple audio signals using a trained model based on the sensing output, one or more first sound sources in one or more desired directions and one or more second sound sources in one or more undesired directions to generate a beamformed output signal originated from the one or more first sound sources.

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

Abstract: Disclosed is an apparatus for immersive spatial audio modeling and rendering for effectively transmitting and playing immersive spatial audio content. The apparatus for immersive spatial audio modeling and rendering disclosed herein may model a spatial audio scene, generate and transmit parameters necessary for spatial audio rendering, and generate various spatial audio effects using the spatial audio parameters, to provide an immersive three-dimensional (3D) audio source coinciding with visual experience in a virtual reality space in response to free changes in the position and direction of a remote user in the space.

Abstract: Provided is an electronic device that includes a communication module, a memory, and a processor operatively connected to the communication module and the memory, which includes instructions that, when executed, cause the processor to perform beamforming training, control to transmit or receive through the communication module a radio signal for a radar function based on the beamforming training, to detect at least one object positioned based on the radio signal transmitted or received in the determined beam direction, to set at least one of the at least one object as a reference object, to periodically transmit or receive the radio signal in a direction of the reference object, to monitor a state of the reference object, based on the radio signal transmitted or received in the direction of the reference object, and to repeat the performing of the beamforming training, based on the monitoring result.

Abstract: The technology described in this document can be embodied in a method that includes receiving an input signal captured by one or more sensors associated with an active noise reduction (ANR) headphone, and determining one or more characteristics of a first portion of the input signal. Based on the one or more characteristics of the first portion of the input signal, a gain of a variable gain amplifier (VGA) disposed in an ANR signal flow path can be adjusted, and accordingly, a set of coefficients for a tunable digital filter disposed in the ANR signal flow path can be selected. The method further includes processing a second portion of the input signal in the ANR signal flow path using the adjusted gain and selected set of coefficients to generate a second output signal for the electroacoustic transducer of the ANR headphone.

Abstract: This disclosure describes a wide band array that uses wideband beamforming with interference cancellation at multiple independent frequencies and spatial locations and main lobe steering at multiple independent frequencies and spatial locations. One embodiment uses one through N analysis filter bands 1410 coupled to one through N narrowband beamformers 1414 with the output processed through one through N synthesis bands and summed together to produce the full spectrum output signal 1426. Another embodiment uses one through M sensors with Discrete Fourier Transforms (DFT) and one through N frequency bins 1412 coupled to one through N narrowband beamformers 1414 processed through an Inverse DFT to produce the full spectrum output signal 1428. Another embodiment uses one through N sensor subarrays and one through N frequency bands 1406 coupled to one through N narrowband beamformers 1414 summed together to produce the full spectrum output signal 1422.

Abstract: Methods, apparatus and techniques for acquiring output signals associated with different sources (such as audio sources) are presented. A first input signal is combined with a delayed and scaled version of a second input signal, to acquire a first output signal. A second input signal is combined with a delayed and scaled version of the first input signal, to acquire a second output signal. Using a random direction optimization, scaling values (forming a candidates' vector) are determined by iteratively modifying the candidates' vector. A Kullback-Leibler divergence is measured. The first output signal and the second output signal are selected to be those measurements associated with the candidate parameters associated with Kullback-Leibler divergence which indicates lowest similarity.

Abstract: An information processing apparatus receives a user operation for issuing an instruction for reproducing a trace recording including information about (a) at least one of a pan value, a tilt value, a zoom value, and a value related to image quality conditions in a predetermined period for a first image capturing unit and (b) tally information indicating a state of video delivery in the predetermined period of the first image capturing unit, controls at least one of the pan value, the tilt value, the zoom value, and the value related to image quality conditions for the first image capturing unit according to the trace recording during a period in a case where the trace recording is being reproduced according to the received user operation, and displays information indicating the state of video delivery of the first image capturing unit according to the tally information recorded in the trace recording.

Abstract: A system configured to perform audio type detection for sound source localization (SSL) data is provided. A device processes audio data representing sounds from multiple sound sources to determine SSL data that distinguishes between each of the sound sources. To identify an audio type associated with a sound source and/or track individual sound sources over time, the device can determine a correlation between the SSL data and an audio event. Examples of audio events include a wakeword component detecting a wakeword or an acoustic event detector detecting a particular acoustic event. The device may determine a correlation between wakeword data indicating that a wakeword is represented in the audio data and SSL data for each individual sound source. The device may then identify a sound source that is most strongly correlated to the wakeword data and associate the wakeword and a corresponding voice command with the sound source.

Abstract: Some implementations relate to methods, systems, and computer-readable media for providing audio for virtual experiences. In some implementations, a method includes receiving, at a server, a first request to generate a plurality of sounds for a user device, wherein the user device is associated with a virtual experience hosted by the server, obtaining, by the server, sound source data for a plurality of sound sources associated with the plurality of sounds, obtaining, by the server, virtual experience state information that comprises a location of a virtual microphone in the virtual experience and at least one of a velocity of the virtual microphone in the virtual experience or an orientation of the virtual microphone in the virtual experience, generating, by the server, an audio mix of the plurality of sounds based on the sound source data and the virtual experience state information, and transmitting the audio mix to the user device.

Abstract: A microphone system is disclosed, comprising: a microphone array and a processing unit. The microphone array comprises Q microphones that detect sound and generate Q audio signals. The processing unit is configured to perform operations comprising: spatial filtering over the Q audio signals using a trained model based on at least one target beam area (TBA) and coordinates of the Q microphones to generate a beamformed output signal originated from ? target sound source inside the at least one TBA, where ?>=0. Each TBA is defined by r time delay ranges for r combinations of two microphones out of the Q microphones, where Q>=3 and r>=1. A dimension of a first number for locations of all sound sources able to be distinguished by the processing unit increases as a dimension of a second number for a geometry formed by the Q microphones increases.

Abstract: An apparatus and method for monitoring audio output is disclosed. The apparatus may include circuitry configured for providing one or more primary audio signals based on signals from one or more first microphones associated with an audio capture device and providing one or more secondary audio signals based on signals from one or more second microphones associated with an audio monitoring device, the audio monitoring device being separate from the audio capture device and configured for output of the one or more primary audio signals and the one or more secondary audio signals through one or more loudspeakers. The apparatus may include circuitry configured for modifying one or both of the primary and secondary audio signals such that output of the one or more primary audio signals are distinguished over output of the one or more secondary audio signals.

Abstract: Disclosed are a method for controlling a sound box, a sound box and a non-transitory computer readable storage medium. The method is applied to the sound box in a sound box system. The sound box system includes at least two sound boxes, each sound box includes at least two first sensors, the at least two first sensors are provided on opposite sides of the sound box, each first sensor is configured to transmit and receive preset signals, and the method includes following operations: obtaining, by the sound box, a connection status with other sound boxes; when the sound box is wirelessly connected to the other sound boxes, obtaining first sensing data detected by a first sensor; determining target channel information according to the first sensing data; and outputting audio according to the target channel information.

Abstract: A camera with an accumulated water treatment function and a method of treating accumulated water thereof are disclosed. The camera includes a main body, a processing module, an image capturing module, a speaker, a first microphone, and a second microphone. The processing module has an image processing unit and an audio processing unit. The image capturing module is used for capturing an environmental image. The first microphone and the speaker are arranged on different sides of the main body. The second microphone and the first microphone are arranged on different sides of the main body. The audio processing unit controls the speaker to broadcast a first audio file and uses the first microphone or the second microphone to record to generate a second audio file so as to determine the difference between the second audio file and a first preset audio file.

Abstract: A method and apparatus for capturing audio including a ceiling mountable second-order differential microphone module. The module including a solid planar baffle having a generally centered aperture, at least one mounting foot to suspend the solid planar baffle approximately parallel with a rear reflecting plane and to space the solid planar baffle at a predetermined distance below the rear reflecting plane, a differential microphone sealably coupled to the planar baffle with a first side of the differential microphone acoustically exposed to an area above the planar baffle and a second side of the differential microphone acoustically exposed to an area below the planar baffle, and a mounting means for mounting the solid rear reflecting panel to the room ceiling.

Abstract: Methods and apparatus to detect spillover are disclosed. An example apparatus includes at least one memory, instructions in the apparatus, and processor circuitry to execute the instructions to: identify a quantity of first durations of loudness in an audio signal of media; calculate a ratio of the quantity of the first durations of loudness to a quantity of second durations of loudness in the audio signal of the media, the quantity of the second durations of loudness including the quantity of the first durations of loudness; and in response to a detection of the audio signal being spillover, store data denoting the media as un-usable to credit a media exposure when the ratio does not satisfy a loudness ratio threshold, the storing of the data to improve an accuracy of media exposure credits by not crediting spillover media.

Abstract: Disclosed are techniques for an improved method for performing sound source localization (SSL) to determine a direction of arrival of an audible sound using a combination of timing information and amplitude information. For example, a device may decompose an observed sound field into directional components, then estimate a time-delay likelihood value and an energy-based likelihood value for each of the directional components. Using a combination of these likelihood values, the device can determine the direction of arrival corresponding to a maximum likelihood value. In some examples, the device may perform Acoustic Wave Decomposition processing to determine the directional components. In order to reduce a processing consumption associated with performing AWD processing, the device splits this process into two phases: a search phase that selects a subset of a device dictionary to reduce a complexity, and a decomposition phase that solves an optimization problem using the subset of the device dictionary.

Abstract: To provide an audio processing device including a configuration in which a different microphone is used for generation of directivity depending on a frequency band.

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

Abstract: A method and apparatus are provided for adaptively generating a directional output signal from sound received by at least two microphones arranged as microphone array.

Abstract: An electronic device and method for generating an audio signal is provided. According to various example embodiments, an electronic device may include a display configured to depict visual information to the outside of the electronic device. The electronic device further includes an actuator to cause the display to vibrate. A processor is electrically connected to the actuator and the display. The processor applies a pilot signal to the actuator, and detects an amount of vibration of the display caused by applying the pilot signal. The processor sets an environment of the audio signal based on the amount of vibration of the display.

Abstract: An ear-worn electronic device comprises a microphone arrangement configured to sense sound in an acoustic environment, an acoustic transducer, and a non-volatile memory configured to store parameter value sets each associated with a different acoustic environment, at least one of which is associated with an acoustic environment with muffled speech. A control input of the device is configured to receive a control input signal produced by a user-actuatable control, a sensor or an external electronic device. A processor is configured to classify the acoustic environment as one with muffled speech using the sensed sound and, in response to a signal received from the control input, apply one or more of the parameter value sets appropriate for the classification to enhance intelligibility of muffled speech.

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

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

Abstract: Presented herein are techniques to generate a stereo sound signal based on direction of arrival of sound signals. A method includes receiving sound signals at a microphone, outputting a mono sound signal from the microphone, determining a direction of arrival of the sound signals with respect to the microphone, generating, from the mono audio signal, a stereo sound signal having a first channel and a second channel, wherein an amplitude of the first channel and an amplitude of the second channel are weighted, respectively, based on the direction of arrival of the sound signals at the microphone.

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

Abstract: A controller includes processor and memory. The memory stores (i) a captured image outputted from a camera in an unmanned aircraft and (ii) sound outputted from a directional microphone in the unmanned aircraft, and stores (i) a position and an orientation of the camera and (ii) a position and an orientation of the directional microphone. The processor obtains, from the memory, the captured image, the sound, the position and the orientation of the camera, and the position and the orientation of the directional microphone, calculates a sound pickup direction of the directional microphone using the position and the orientation of the camera and the position and the orientation of the directional microphone, superimposes an object indicating the sound pickup direction at a position, on the captured image, corresponding to the sound pickup direction calculated, and causes a display device to display the captured image on which the object is superimposed.

Abstract: Techniques are described herein for providing an interactive interface for live streaming events. A user may view a broadcasted live streaming event via the interactive interface. A sub-group of participants of the live streaming event may be identified from a larger group of participants based at least in part on respective locations of respective computing devices of the sub-group participants. A graphical representation can be generated for the group using images of each of the sub-group participants. A selection of the sub-group can be received from the interactive interface and the graphical representation can be presented in response to receiving the selection. Various features of the interactive interface provide for a more enriching experience for both the performer(s) and viewing participants of the broadcasted live streaming event.

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

Abstract: Method, apparatus, and computer-readable media focusing sound signals from plural microphones in a 3D space, to determine audio signal processing profiles to optimize sound source(s) in the space. At least one processor determines plural virtual microphone bubbles in the space, and defines one or more bubble object profiles which comprise(s) specific attributes and functions of audio processing functions for each bubble, each bubble object profile including: (a) an individual bubble object profile when the bubble has been configured for an individual bubble; (b) a region object profile when the bubble has been configured for a region of one or more bubbles; and (c) a group object profile when the bubble has been configured for a group having one or more bubbles. The audio signal processing functions are used for the at least one bubble, for any combination of (a), (b), and (c).

Abstract: A method of tracking sound sources in an environment by monitoring the local area with a primary sensor system comprising a microphone array. The location of sound sources in the local area are tracked using the monitored sound. If an ambiguity is determined in the tracked location of the sound source, a secondary sensor system is activated. The secondary sensor system has a larger power draw than the primary sensor system. The secondary sensor system determines the updated location of the sound source. The tracked location of the sound sources is updated to be the updated location. Once the location is updated, the secondary sensor system is deactivated.

Abstract: Disclosed are a microphone array-based sound signal processing method, apparatus and device. The method comprises: selecting, from a microphone array, a target microphone combination which is used for receiving a sound signal of a target sound source, the target microphone combination comprising a first microphone and at least one second microphone; obtaining target compensation information corresponding to the target microphone combination, the target compensation information comprising a signal compensation parameter of each second microphone with respect to the first microphone; according to the target compensation information, performing signal compensation processing on a second sound signal received by means of the second microphone; and according to the second sound signal subjected to the signal compensation processing and a first sound signal received by means of the first microphone, obtaining a target sound signal and outputting same.

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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