Zweiwegelautsprecher mit schwebendem Wellenleiter

Eine oder mehrere Ausführungsformen der vorliegenden Offenbarung betreffen eine Zweiwegelautsprecherauslegung, die eine kondensierte Geometrie zwischen Niederfrequenz-(LF) und Hochfrequenz- (HF)-Treibern erzwingt und dann einen Mitteltonwellenleiter vor dem LF-Treiber „schweben“ lässt. Dies ist eine Hybridauslegung, die von der großen Nähe akustischer Mittelpunkte profitieren soll, ohne ein Mittelachsenhindernis für den LF-Treiber einzubringen. Außerdem werden die LF- und HF-Wellenleiter und zugehörige akustische Elemente verwendet, um sehr niederfrequente Energie, die von dem LF-Wellenleiter nicht ausreichend unterstützt wird, so umzuleiten, dass sie über andere Wege ungehindert austritt.

DEVICE AND PROCEDURE FOR THE CONTROL OF A PUBLIC-ADDRESS SYSTEM AND PUBLIC-ADDRESS SYSTEM

ARRANGEMENT FOR THE CLAY/TONE ADMISSION AND TONWIEDERGABE OF MEANS OF A MULTIPLICITY OF SENSORS

HOUSING FOR MICROPHONE ARRAYS AND MULTI-SENSOR ARRANGEMENTS FOR YOUR GRÖßEN OPTIMIZATION

CIRCULARLY SYMMETRIC, ZERO REDUNDANCY, PLANAR ARRAY HAVING BROAD FREQUENCY RANGE APPLICATIONS

A class of planar arrays having broad frequency range applications for sour ce location, source imaging or target illumination with projected beams is described in this disclosure. The non-redundant arrays are circularly symmetric and made up of a plurality of sensing and/or transmitting elements arranged so as to substantially eliminate grating lobes for a broad range of frequencies. Signals received from or transmitted to the elements are appropriately phased to control the beam of the array. ...

СПОСОБ И СИСТЕМА ДЛЯ НАПРАВЛЕННОГО ЗАХВАТА АУДИОСИГНАЛА

Способ и система для цифрового направленного фокусирования и управления направлением дискретизированного аудиосигнала в целевой области для получения избирательного выхода аудиосигнала, сопровождающего видеосигнал. В предпочтительном варианте осуществления способ и система отличаются тем, что получают данные расположения и фокусировки от одной или более камер, снимающих событие, и используют эти вводимые данные для формирования соответствующего выхода аудиосигнала совместно с изображением.

Array microphone system and assembling method thereof

Bidirectional loudspeaker with floating waveguide

SYSTEM OF SOUND RECORDING SELECTIVE FOR ENVIRONMENT REVERBERATING AND NOISY

L'invention concerne un système de prise de son sélective de type antenne acoustique.Il comprend un aérien (1) formé par une partie de surface cylindrique concave (S), comportant une pluralité de transducteurs électroacoustiques (M0 à M10 ) répartis sur et au voisinage de la surface cylindrique (S). Des circuits de sommation partielle (21 à 24 ) d'une pluralité de signaux analogiques sonores permettent de former des antennes acoustiques élémentaires et un circuit de sommation (5) permet de délivrer un signal analogique résultant représentatif du signal sonore issu d'une zone utile de locution (L).Application à la téléphonie mains-libres, aux stations de travail multimédia. The invention relates to an acoustic antenna type selective sound pickup system comprising an aerial (1) formed by a concave cylindrical surface portion (S), comprising a plurality of electroacoustic transducers (M0 to M10) distributed over and over vicinity of the cylindrical surface (S). Partial summation circuits (21 to 24) of a plurality of analog sound signals make it possible to form elementary acoustic antennas and a summation circuit (5) makes it possible to deliver a resulting analog signal representative of the sound signal coming from a useful area of speech (L). Application to hands-free telephony, multimedia workstations.

MICROPHONE ARRAY FOR ACOUSTIC SOURCE SEPARATION

A system for directionally selective sound reception comprises an array of pressure sensors (120a, 120c) each arranged to output a pressure signal indicative of pressure, and a processor arranged to receive the pressure signals. The sensor array comprises a support (130)supporting the four sensors. Two of the sensors are mounted on one side of the support and at least a third sensor is supported on an opposite side of the support. The sound pressure difference measured between the first sensor and the second sensor caused by sound arriving at the array from a direction parallel to the support (130) is dependent on the distance between the first and second sensors and the nature of material in the space between the first and second sensors. The sound pressure difference measured between the first and third sensors caused by sound travelling perpendicular to the support is dependent on the distance between the first and third sensors. The nature of material in the space between the first ...

Circularly symmetric, zero redundancy, planar array having broad frequency range applications

A class of planar arrays having broad frequency range applications for source location, source imaging or target illumination with projected beams is described in this disclosure. The non-redundant arrays are circularly symmetric and made up of a plurality of sensing and/or transmitting elements arranged so as to substantially eliminate grating lobes for a broad range of frequencies. Signals received from or transmitted to the elements are appropriately phased to control the beam of the array.

Multi-channel multi-domain source identification and tracking

An audio source location, tracking and isolation system, particularly suited for use with person-mounted microphone arrays. The system increases capabilities by reducing resources required for certain functions so those resources can be utilized for result enhancing processes. A wide area scan may be utilized to identify the general vicinity of an audio source and a narrow scan to locate pinpoint positions may be initiated in the general vicinity identified by the wide area scan. Subsequent locations may be anticipated by compensating for motion of the sensor array and anticipated changes in source location by trajectory. Identification may use two or more sets of characterizations and rules. The characterizations may use computationally less intense analyses to characterize audio and only perform computationally higher intensity analysis if needed. Rule sets may be used to eliminate the need to track audio sources that emit audio to be eliminated from an audio output.

MULTI-CHANNEL MULTI-DOMAIN SOURCE IDENTIFICATION AND TRACKING

An audio source location, tracking and isolation system, particularly suited for use with person-mounted microphone arrays. The system increases capabilities by reducing resources required for certain functions so those resources can be utilized for result enhancing processes. A wide area scan may be utilized to identify the general vicinity of an audio source and a narrow scan to locate pinpoint positions may be initiated in the general vicinity identified by the wide area scan. Subsequent locations may be anticipated by compensating for motion of the sensor array and anticipated changes in source location by trajectory. Identification may use two or more sets of characterizations and rules. The characterizations may use computationally less intense analyses to characterize audio and only perform computationally higher intensity analysis if needed. Rule sets may be used to eliminate the need to track audio sources that emit audio to be eliminated from an audio output. 1. An audio processing system comprising:a body mounted microphone array;an accelerometer linked to said microphone array;an audio source locating unit connected to said microphone array having an output representative of a location of an audio source;a location table connected to said output of said audio source locating unit containing a representation of a location of one or more audio sources; andan array displacement compensation unit having an input connected to an output of said accelerometer and an output representative of a change in position of said accelerometer, wherein said location table is responsive to said output representative of a change in position of said accelerometer to update the representation of said one or more audio sources to compensate for said change in position of said accelerometer.2. An audio processing system according to further comprising a localized audio capture unit connected to said microphone array and said location table to capture and isolate audio ...

VOICE CONFERENCE DEVICE

Microphones arranged in an array shape along a longitudinal direction are respectively formed in both the longitudinal side surfaces of a housing 2 with substantially an elongated rectangular parallelepiped shape, and speakers arranged in an array shape along the longitudinal direction are formed in a lower surface. The speaker array forms sound emission beams based on sound emission directivity set according to a conference environment. On the other hand, when the microphone array forms sound collection beams by sound collection signals collected, a talker direction is detected from these beams and an output sound signal corresponding to this direction is formed and also is reflected on setting of the sound emission directivity. Also, when there are plural input sound signals, the sound emission directivity is set according to a use situation of the plural input sound signals.

LINEAR FILTERING FOR NOISE-SUPPRESSED SPEECH DETECTION VIA MULTIPLE NETWORK MICROPHONE DEVICES

Wide array of circularly symmetric zero-redundancy planes over a wide frequency range

SCHALLSTRAHL-LAUTSPRECHERSYSTEM

Loudspeaker bracket

The bracket comprises a frame 400 and a suspension member 402. The suspension member is moveable between a retracted position (fig 4a) and an extended position in which the suspension member projects from the frame and defines an axis of rotation. The suspension member is configured in the extended position for attachment to a frame of a further bracket attached to a further speaker such that the further speaker is rotatable about the axis of rotation (fig 5b). The suspension member is configured in the extended position such that the axis of rotation is between the speaker and the further speaker. The cam 430 may be turned to select the angle of rotation between the loudspeakers, and it acts as an adjustable end stop to limit that rotation.

ELECTROACOUSTIC TRANSDUCER WITH SEVERAL LOUDSPEAKERS

ELECTROACOUSTIC LINE ARRAY TRANSFORMATION

PROCEDURE FOR THE INTOXICATION REDUCTION IN A HEARING AID AND AFTER SUCH PROCEDURES FUNCTIONING HEARING AID

SOUND RADIATION LOUDSPEAKER SYSTEM

СПОСОБ И СИСТЕМА ДЛЯ НАПРАВЛЕННОГО ЗАХВАТА АУДИОСИГНАЛА

Способ и система для цифрового направленного фокусирования и управления направлением дискретизированного аудиосигнала в целевой области для получения избирательного выхода аудиосигнала, сопровождающего видеосигнал. В предпочтительном варианте осуществления способ и система отличаются тем, что получают данные расположения и фокусировки от одной или более камер, снимающих событие, и используют эти вводимые данные для формирования соответствующего выхода аудиосигнала совместно с изображением.

DEVICE OF SEIZURE AND RESTITUTION OF THE SOUND USING SEVERAL SENSORS

SOUND BEAM LOUDSPEAKER SYSTEM

A loudspeaker system is described including an array of electro-acoustic transducers capable of generating steerable beams of sound and additional transducers adapted to reproduce low frequency sound being placed at the perimeter of said array.

System for selective sound capture for reverberant and noisy environment

A system for selective sound capture for hands-free telephony and multimedia workstations of the acoustic antenna type with an aerial formed by a member with a concave cylindrical surface. The system includes a plurality of electro-acoustic transducers which are distributed over and in the vicinity of the cylindrical surface. Circuits for partial summation of a plurality of analog sound signals make it possible to form elementary acoustic antennas and a summation circuit produces a resultant analog signal representative of the sound signal arising from a useful speaking area.

Microphone array with automated adaptive beam tracking

An example method of operation may include initializing a microphone array in a defined space to receive one or more sound instances based on a preliminary beamform tracking configuration, detecting the one or more sound instances within the defined space via the microphone array, modifying the preliminary beamform tracking configuration, based on a location of the one or more sound instances, to create a modified beamform tracking configuration, and saving the modified beamform tracking configuration in a memory of a microphone array controller.

Personal audio device

A personal audio device configured to be worn on the head or body of a user and including a plurality of microphones configured to provide a plurality of separate microphone signals capturing audio from an environment external to the personal audio device, and a processor configured to process a first subset of the plurality of separate microphone signals using a first array processing technique to provide a first array signal, compare the first array signal to a microphone signal from the plurality of separate microphone signals, and select the first array signal or the microphone signal based on the comparison.

COMPACT SOUND LOCATION MICROPHONE

A system, method and program product for improved techniques for sound management and sound localization is provided. The present invention provides for improving sound localization and detection by inputting a predetermined location's dimensional data and location reference and processing detected sound details, detection device details and the associated location dimensional data as sound localization information for multi-dimensional display. The present invention provides mapping information of sound, people and structural information for use in multiple applications including residential, commercial and emergency situations. 1. A method for defining a reference sound position and producing an animated indicia responsive to one or more sound characteristics , comprising:generating the animated indicia responsive to sound;defining at least one sound characteristic to be detected;detecting at least one target sound in a three dimensional physical space in relation to the at least one sound characteristic;determining the referenced sound position in relation to the detected target sound in the three dimensional physical space,wherein the animated indicia responds proximate to the determined referenced sound position, andwherein orientation of the animated indicia changes in relation to orientation of the detected target sound in the three dimensional physical space.2. The method of claim 1 , wherein the sound characteristics are one or more of: frequency range claim 1 , decibel level claim 1 , pitch range claim 1 , loudness range and directional location.3. The method of claim 2 , wherein the referenced sound position includes a two-dimensional position in relation to the predetermined location.4. The method of claim 3 , wherein the referenced sound position includes a three-dimensional position in relation to a predetermined location.5. The method of claim 2 , wherein the at least one target sound is detected using one or more of a soundwave detector claim 2 , ...

SYSTEMS, METHODS, AND APPARATUS FOR ENHANCED CREATION OF AN ACOUSTIC IMAGE SPACE

СИСТЕМА ОБРАБОТКИ ИНФОРМАЦИИ И НОСИТЕЛЬ ИНФОРМАЦИИ

Изобретение относится к области обработки информации. Техническим результатом является расширение арсенала технических средств для обработки информации. Система обработки информации включает в себя: модуль распознавания, выполненный с возможностью распознавания первой цели, расположенной в первом месте, на основе множества первых сигналов, детектируемых множеством первых датчиков, размещенных вокруг конкретного пользователя, расположенного во втором месте; модуль идентификации, выполненный с возможностью идентификации первой цели, распознанной модулем распознавания; модуль оценки, выполненный с возможностью оценивать положение конкретного пользователя на основе первого сигнала, детектируемого одним из множества первых датчиков; и модуль обработки сигналов, выполненный с возможностью: модификации каждого из множества вторых сигналов, полученных из множества вторых датчиков, размещенных вокруг первой цели, на основе параметра акустической информации третьего места, отличного от первого и ...

Elektroakustischer Wandler mit mehreren Lautsprechern

Redundanzlose kreissymmetrische ebene Gruppenantenne

Optimal modal beamformer for sensor arrays

Elimination of microphone feedback using cardioid antennas

Loudspeaker bracket

LOUDSPEAKER DEVICE WITH STEERED DIRECTION SENSITIVITY

Methods circuits devices systems and associated computer executable code for acquiring acoustic signals

Disclosed are methods, circuits, devices, systems and associated computer executable code for acquiring, processing and rendering acoustic signals. According to some embodiments, one or more direction specific audio signals may be generated using a microphone array comprising two or more microphones and an audio stream generator. The audio stream generator may receive a direction parameter from an optical tracking system. There may be provided an audio rendering system adapted to normalize and/or balance acoustic signals acquired from a soundscape.

AUTOMATED TRANSCRIPT GENERATION FROM MULTI-CHANNEL AUDIO

Systems and methods are described for generating a transcript of a legal proceeding or other multi-speaker conversation or performance in real time or near-real time using multi-channel audio capture. Different speakers or participants in a conversation may each be assigned a separate microphone that is placed in proximity to the given speaker, where each audio channel includes audio captured by a different microphone. Filters may be applied to isolate each channel to include speech utterances of a different speaker, and these filtered channels of audio data may then be processed in parallel to generate speech-to-text results that are interleaved to form a generated transcript.

DEVICE FOR MONITORING THE USE OF BLISTER PACKAGED CONTENTS AT A DISTANCE

There is provided a device for monitoring the use of a blister package, strip package, vial or bottle contents at a distance. A processor is connected to a compact random or quasi-random n-microphone array and is programmed to detect the sound of the content being expelled from a blister cavity, strip package, or a cap being removed from a vial or bottle. A content use data memory associated with the processor stores information relating to the expulsion or removal events. The processor is equipped with statistical means for differentiating the sound of the content being expelled, from the background noise, generating an electrical signal that is analyzed for relevance to content use events by the processor, and storing the resulting use data in memory. The processor may have an adaptive beam focussing algorithm to determine the direction of the source of the sound.

DEVICE FOR MONITORING THE USE OF BLISTER PACKAGED CONTENTS AT A DISTANCE

There is provided a device for monitoring the use of a blister package, strip package, vial or bottle contents at a distance. A processor is connected to a compact random or quasi-random n-microphone array and is programmed to detect the sound of the content being expelled from a blister cavity, strip package, or a cap being removed from a vial or bottle. A content use data memory associated with the processor stores information relating to the expulsion or removal events. The processor is equipped with statistical means for differentiating the sound of the content being expelled, from the background noise, generating an electrical signal that is analyzed for relevance to content use events by the processor, and storing the resulting use data in memory. The processor may have an adaptive beam focussing algorithm to determine the direction of the source of the sound.

DEVICE OF SEIZURE AND RESTITUTION OF THE SOUND USING SEVERAL SENSORS

La présente invention a pour objet un système de diffusion d'un signal sonore comprenant une pluralité de diffuseurs sonores et des moyens de traitement du signal électrique délivré par chacun desdits diffuseurs, comportant un amplificateur, un circuit de conversion analogique-numérique, un circuit de filtrage et un circuit de sommation des signaux filtrés, pour délivrer au moins un signal de sortie, caractérisé en ce que les diffuseurs sonores sont répartis selon au moins un axe, avec un pas logarithmique.

Method and apparatus of recording and multiple reproduction of the sounds

MICROPHONE ARRAY STRUCTURE, METHOD AND APPARATUS FOR FORMING BEAM HAVING CONSTANT REFLECTIVITY AND METHOD AND APPARATUS FOR ESTIMATING DIRECTION OF SOUND SOURCE EMPLOYING THE SAME

PURPOSE: A microphone array structure, a method and an apparatus for forming a beam having a constant reflectivity and a method and an apparatus for estimating the direction of a sound source employing the same are provided to improve the recognition rate of the sound. CONSTITUTION: An apparatus for forming a beam having a constant reflectivity by employing the microphone array structure includes a microphone array(211), a beam forming unit(231), a filtering unit(251) and an adding unit(271). The microphone array(211) is provided with a first sub-array(213), a second sub-array(215) and a third sub-array(215). The first sub-array(213) is placed at a predetermined position of the flat plate. And, the second sub-array(215) and the third sub-array(217) are located at the position corresponding to the first to ith predetermined distance vertical to the extending line of the first sub-array(213) and the central axis of the flat plate in response to the first to ith target frequencies of each ...

WIND NOISE REJECTION APPARATUS

An apparatus for reduction of wind noise comprised of an electro-acoustic transducer arrangement with at least one transducer element. The exposed structure is covered with at least one thin layer of wind-resistive material, in the form of a mesh, and optionally includes one or more further layers of felt, foam and/or mesh in any combination. Where a plurality of elements is provided, the electrical outputs of the elements may be added together to provide an output signal with increased signal to wind noise ratio. The signal may be subject to additional signal processing such as filtering and/or level sensitive signal inhibition. COPYRIGHT KIPO & WIPO 2010 ...

METHODS, APPARATUS, AND COMPUTER-READABLE MEDIA FOR ORIENTATION-SENSITIVE RECORDING CONTROL

Wind noise rejection apparatus

An apparatus for reduction of wind noise comprised of an electro-acoustic transducer arrangement with at least one transducer element. The exposed structure is covered with at least one thin layer of wind-resistive material, in the form of a mesh, and optionally includes one or more further layers of felt, foam and/or mesh in any combination. Where a plurality of elements is provided, the electrical outputs of the elements may be added together to provide an output signal with increased signal to wind noise ratio. The signal may be subject to additional signal processing such as filtering and/or level sensitive signal inhibition.

HOUSING FOR MICROPHONE ARRAYS AND MULTI-SENSOR DEVICES FOR THEIR SLZE OPTIMIZATION

A Sensor System (101) being located in an environment composed of a first medium (105), where waves propagate with a first phase velocity, the sensor System (101) comprising at least one main enclosure (102) and a sensor array (103) with at least two sensors (104), said sensor array (103) being arranged inside the main enclosure (102), wherein the space inside the main enclosure (102) between the sensor array (103) and the inner surface of the main enclosure (102) is filled with a second medium (106), in which waves propagate with a second phase velocity, the second phase velocity being different from the first velocity.

Method for extending the frequency range of a beamformer without spatial aliasing

A conferencing unit, comprising an array of microphones embedded in a diffracting object configured to provide a desired high frequency directivity response at predetermined microphone positions, and a low frequency beamformer operable to achieve a desired low frequency directivity response, wherein the beamformer is linearly constrained to provide a smooth transition between low and high frequency directivity responses.

Audio hub and a system having one or more audio hubs

There may be provided a system that may include a processor and an audio hub; wherein the audio hub may include first communication interfaces, a second communication interface, a processor, and a memory; wherein the first communication interfaces may be configured to exchange audio signals with a group of audio components of different types; wherein an aggregate number of first communication interface bits exceeds a number of second communication interface bits; wherein the audio signals may include input audio signals received from the group and output audio signals transmitted to the group; wherein the processor may be configured to generate an input multiplex of input audio signals; and wherein the second communication interface may be configured to transmit the input multiplex to the processor and to receive an output multiplex from the processor.

Linear Filtering for Noise-Suppressed Speech Detection

Systems and methods for suppressing noise and detecting voice input in a multi-channel audio signal captured by a plurality of microphones include (i) capturing a first audio signal via a first microphone and a second audio signal via a second microphone, wherein the first and second audio signals respectively comprises first and second noise content from a noise source; (ii) identifying the first noise content in the first audio signal; (iii) using the identified first noise content to determine an estimated noise content captured by the plurality of microphones; (iv) using the estimated noise content to suppress the first and second noise content in the first and second audio signals; (v) combining the suppressed first and second audio signals into a third audio signal; and (vi) determining that the third audio signal includes a voice input comprising a wake word.

Microphone arrays and communication systems for directional reception

Disclosed herein are microphone arrays for directional reception, along with related system, devices, and techniques. For example, a four-microphone array for directional signal reception may include first, second, and third microphones arranged such that projections of the first, second, and third microphones in a plane provide corners of a triangle in the plane. In some embodiments, a fourth microphone may be arranged such that a projection of the fourth microphone in the plane is disposed in an interior of the triangle. In other embodiments, the fourth microphone may be arranged such that the projection of the fourth microphone in the plane is disposed outside the interior of the triangle, and a distance between the first microphone and the second microphone is different from a distance between the first microphone and the third microphone.

Acoustic deflector for omni-directional speaker system

An omni-directional acoustic deflector includes an acoustically reflective body that has a truncated conical shape which includes a substantially conical outer surface that is configured to be disposed adjacent an acoustically radiating surface (e.g., a diaphragm) of an acoustic driver thereby to define an acoustic radiation path therebetween. The acoustically reflective body is profiled such that a cross-sectional area of the acoustic radiation path increases monotonically with respect to radial distance from a motion axis of the acoustic driver.

Detecting events based on the rhythm and flow of a property

Methods, systems, and apparatus, including computer programs encoded on a storage device, for a monitoring system that is configured to detect an event at a property. The monitoring system may include a processor and a storage device storing instructions that, when executed by the processor, cause the processor to perform operations. The operations include obtaining current activity data that (i) is generated by monitoring system components and (ii) represents two or more activities that have occurred at the property between a first time and a second time, accessing historical activity data that represents historical activities that have been learned by the monitoring system, determining, by the monitoring system and based on (i) the current activity data and (ii) the historical activity data, whether an event has been detected, and based on determining that an event has been detected, performing one or more operations based on the detected event.

RADIATION FIGURATION TRANSDUCER ARRAY

AUDIO CONFERENCING APPARATUS

Microphones arranged in an array shape along a longitudinal direction are respectively formed in both the longitudinal side surfaces of a housing 2 with substantially an elongated rectangular parallelepiped shape, and speakers arranged in an array shape along the longitudinal direction are formed in a lower surface. The speaker array forms sound emission beams based on sound emission directivity set according to a conference environment. On the other hand, when the microphone array forms sound collection beams by sound collection signals collected, a talker direction is detected from these beams and an output sound signal corresponding to this direction is formed and also is reflected on setting of the sound emission directivity. Also, when there are plural input sound signals, the sound emission directivity is set according to a use situation of the plural input sound signals.

ANTENNA FORMEE Of a PLURALITY OF ACOUSTIC SENSORS

Loudspeaker system

Provided is a loudspeaker system capable of easily determining control parameters for controlling the directional characteristics of a speaker set including two or more speaker units. The loudspeaker system is configured to include: a speaker set 11 including two or more speaker units 2; a camera 15 adapted to photograph an acoustic space of the speaker set 11; a monitor 22 adapted to display a camera image 30 photographed by the camera 15; and a directional control parameter generating part 24 adapted to generate directional control parameters providing the directional characteristics of the speaker set 11. The directional control parameter generating parts 24 includes: a target position designation part 101 adapted to, on the basis of user operations, designate target positions 32 on the camera image 30; a directional control angle calculation part 103 adapted to, on the basis of the target positions 32, obtain directional control angles θ with respect to a front direction of the speaker ...

SOUND EMISSION AND COLLECTION APPARATUS AND CONTROL METHOD OF SOUND EMISSION AND COLLECTION APPARATUS

A level ratio calculation circuit calculates average signal level data of signal level data corresponding to each sound collection beam signal, and calculates a level ratio between the average signal level data and each of the signal level data. Since a diffraction sound is substantially equal to all the signal level data, a diffraction sound component of the average signal level data also becomes substantially equal. On the other hand, a collection sound from a speaker is specific to the signal level data of the corresponding sound collection beam signal. Therefore, at the level ratio, the portion corresponding to the diffraction sound is flat and a data level becomes high locally in only the portion corresponding to the collection sound. By using this, the sound collection beam signal including the collection sound is detected.

Microphone Array System

A microphone array system or microphone array unit for a conference system is provided that includes a front board, side walls and a plurality of microphone capsules arranged in or on the front board mountable on or in a ceiling of a conference room. The microphone array system or unit is adapted for generating a steerable beam within a maximum detection angle range. The microphone array system or microphone array unit includes a processing unit which is configured to receive the output signals of the microphone capsules and to steer the beam based on the received output signal of the microphone array. The processing unit is configured to control the microphone array to limit the detection angle range to exclude at least one predetermined exclusion sector in which a noise source is located. 1. A microphone array system mountable on or in a ceiling of a conference room , the microphone array system comprising:a front board made of a sound-reflecting material, the front board comprising an upper side, a lower side and a plurality of first openings;side walls extending on the upper side of the front board;a plurality of microphone capsules arranged in or on the front board, wherein the microphone capsules are located at the first openings of the front board, and wherein each microphone capsule is sealed against the front board, and wherein the microphone capsules can acquire only sound entering through the first openings of the front board; anda processing unit which is configured to receive output signals of the microphone capsules and to execute audio beam forming based on the received output signals of the microphone capsules for predominantly acquiring sound coming from an audio source in a first direction.2. The microphone array system according to claim 1 , wherein the microphone capsules are attached to the upper side of the front board by an adhesive film.3. The microphone array system according to claim 1 , further comprising a gauze and a cover board mounted ...

CONFERENCE SYSTEM WITH A MICROPHONE ARRAY SYSTEM AND A METHOD OF SPEECH ACQUISITION IN A CONFERENCE SYSTEM

A conference system including a microphone array unit having a plurality of microphone capsules arranged in or on a board mountable on or in a ceiling of a conference room. The microphone array unit has a steerable beam and a maximum detection angle range. The conference system further includes a processing unit which is configured to receive the output signals of the microphone capsules and to steer the beam based on the received output signal of the microphone array unit. The processing unit is configured to control the microphone array to limit the detection angle range to exclude at least one predetermined exclusion sector in which a noise source is located.

MULTI-ELEMENT DIRECTIONAL ACOUSTIC ARRAYS

Voice conference device

AUDIO DEVICE AND METHOD FOR PRODUCING A SOUND FIELD

СИСТЕМА ОБРАБОТКИ ИНФОРМАЦИИ И НОСИТЕЛЬ ИНФОРМАЦИИ

... 1. Система обработки информации, включающая в себя:модуль распознавания, выполненный с возможностью распознавания первой цели и второй цели на основе сигналов, детектируемых множеством датчиков, размещенных вокруг конкретного пользователя;модуль идентификации, выполненный с возможностью идентификации первой цели и второй цели, распознанной модулем распознавания;модуль оценки, выполненный с возможностью оценивать положение конкретного пользователя в соответствии с сигналом, детектируемым любым одним из множества датчиков; имодуль обработки сигналов, выполненный с возможностью обработки каждого из сигналов, полученных из датчиков вокруг первой цели и второй цели, идентифицированной модулем идентификации таким образом, что при его выводе из множества исполнительных элементов, расположенных вокруг конкретного пользователя, сигналы локализуются рядом с положением конкретного пользователя, в соответствии с оценкой, полученной модулем оценки.2. Система обработки информации по п. 1,в которой первая ...

LAUTSPRECHERVORRICHTUNG MIT GESTEUERTER RICHTUNGSEMPFINDLICHKEIT

A microphone wind shield or wind screen

The wind shield 10 reduces wind noise in electro-acoustic transducer arrangements comprising at least one transducer element, preferably a microphone. The microphone(s) BCD are covered with at least one thin layer of wind-resistive material 10 in the form of a mesh having holes less than approximately 125 microns in size, and preferably in the range 40-50 microns. Optionally the wind shield includes one or more further layers of felt, foam and/or mesh in any combination. Where a plurality of microphones is provided, the electrical outputs of the microphones may be added together to provide an output signal with increased signal to wind noise ratio. The signal may be subject to additional signal processing such as filtering and/or level sensitive signal inhibition (figs 2 and 3).

Elimination of microphone feedback using cardioid antennas

Current anti-feedback systems detect feedback and suppress this by either changing the frequency response, adding a phase shift or changing the frequency of the audio. This enables the system to achieve a higher loop gain giving a higher output for the loudspeaker. This system differs from this by determining the maximum loop gain before feedback in a setup phase. The loop gain of each microphone combination is limited below this level. So, this system does not increase the maximum output but limits it below a predetermined level. This system calculates the loop gain by determining the effective audio path length of each microphone in respect to each loudspeaker in real time by measuring the RF signal strength as received at the loudspeaker from the microphone, as well as gain and frequency response of the system. This system can be used in conjunction with existing anti-feedback systems that enable a higher output from the loudspeaker.

Micophone mixing for wind noise reduction

Signal processing device for acoustic transducer array

METHOD FOR EXTENDING THE FREQUENCY RANGE OF A BEAMFORMER WITHOUT SPATIAL ALIASING

A conferencing unit, comprising an array of microphones embedded in a diffracting object configured to provide a desired high frequency directivity response at predetermined microphone positions, and a low frequency beamformer operable to achieve a desired low frequency directivity response, wherein the beamformer is linearly constrained to provide a smooth transition between low and high frequency directivity responses.

DUAL CONFIGURATION SPEAKER

A compact and portable loudspeaker system operable in two configurations includes a driver array (150) and a base unit (110) having a bass enclosure and a dock. In an extended configuration, the driver array (150) is supported by one or more extension legs (160) between three to eight feet above the base unit (110) where one of the extension legs is held by the dock. The base unit may be placed on a floor of a small to medium venue with the mid-to-high range driver array elevated near or above the elevation of an audience in the venue. In a compact configuration, the driver array (150) is directly supported by the dock and the base unit (110) and driver array (150) may be placed on a table or desk in a classroom, conference room, or other such small to medium venue. The portable loudspeaker system may be transported in the compact configuration.

SPEAKER MODULE, ELECTRONIC DEVICE INCLUDING THE SPEAKER MODULE, AND DISPLAY DEVICE INCLUDING THE ELECTRONIC DEVICE

ACOUSTIC ANTENNA FOR DATA-PROCESSING WORK STATION

A SPEAKER SYSTEM WHICH COMPRISES SPEAKER DRIVER GROUPS

A speaker system (1, 1', 1'') comprises a first speaker driver group (G1), comprising at least one first speaker driver (S1) and a first amplifier (A1), and a second speaker driver group (G2), comprising at least one second speaker driver (S2) and a second amplifier (A2). The speaker system further comprises a digital signal processor (2), adapted to provide a first signal to the first speaker driver group (G1) and a second signal to the second signal group (G2), wherein the first and second signals differ with respect to frequency range and at least one of the speaker driver groups comprises at least 4, 6, 8 or 10 speaker drivers.

HELMET-MOUNTED MULTI-DIRECTIONAL SENSOR

A multi-directional sensor may include a microphone array of three or more microphones mounted on a protective headgear. The microphone array may be positioned and configured so that its far field azimuth sensing range is unobstructed by the protective headgear. An accelerometer may be provided and mounted in a location which is fixed with respect to the microphones of the microphone array. A beacon, such as an ultrasonic transmitter or BLE (Bluetooth Low Energy) transmitter may be associated with or attached to the protective headgear. The microphone array may be utilized with a beam-forming system in order to determine location of an audio source and a beam-steering system in order to isolate audio emanating from the direction of the audio source. The beam-forming system is suitable for tracking the movement of the audio source in order to inform the beam-steering system of the direction or location to be isolated. Because the microphone array will move with a user, a motion sensor may be provided to reduce the computational resources required for tracking and isolation by allowing compensation for change in position and orientation of the user. The beacon will facilitate location of the wearer. 1. A body-mounted multi-directional sensor comprising:a base configured to be worn by a user;a multi-directional acoustic sensor mounted on said base and wherein said base is a protective headgear.2. A body-mounted multi-directional sensor wherein said multi-directional acoustic sensor further comprised three or more microphones mounted in a configuration with a first microphone mounted in a position that is not co-linear with a second microphone and a third microphone.3. A body-mounted multi-directional sensor according to further comprising a fourth microphone mounted in a location that is not co-planar with said first microphone claim 2 , said second microphone and said third microphone.4. A multi-directional sensor according to wherein said microphones are mounted on ...

Microphone

A microphone comprises: a vibration member having a plate shape and formed so as to have elasticity and cause bending due to sound waves; a case having a depressed groove, which forms an air layer between the depressed groove and the vibration member since the vibration member is stacked on an upper part thereof, and coupled to the upper part of the depressed groove such that the vibration member is vibrated by sound waves; and a vibration detection unit provided on the vibration unit and the case and measuring the vibration of the vibration member vibrating by sound waves on the case, wherein when the vibration member vibrates, the air in the air layer flows so as to damp the vibration of the vibration member.

ASYNCHRONOUS AD-HOC DISTRIBUTED MICROPHONE ARRAY PROCESSING IN SMART HOME APPLICATIONS USING VOICE BIOMETRICS

Voice biometrics scoring is performed on received asynchronous audio outputs from microphones distributed at ad hoc locations to generate confidence scores that indicate a likelihood of an enrolled user speech utterance in the output, a subset of the outputs is selected based on the confidence scores, and the subset is spatially processed to provide audio output for voice application use. Alternatively, asynchronous spatially processed audio outputs and corresponding biometric identifiers are received from corresponding devices distributed at ad hoc locations, audio frames of the outputs are synchronized using the biometric identifiers, and the synchronized frames are coherently combined. Alternatively, uttered speech associated with respective ad hoc distributed devices is received and non-coherently combined to generate a final output of uttered speech. The uttered speech is recognized from respective spatially processed outputs generated by the respective devices using biometrics of ...

METHOD TO DETERMINE LOUDSPEAKER CHANGE OF PLACEMENT

A system and method is described for determining whether a loudspeaker device has relocated, tilted, rotated, or changed environment such that one or more parameters for driving the loudspeaker may be modified and/or a complete reconfiguration of the loudspeaker system may be performed. In one embodiment, the system may include a set of sensors. The sensors provide readings that are analyzed to determine 1) whether the loudspeaker has moved since a previous analysis and/or 2) a distance of movement and/or a degree change in orientation of the loudspeaker since the previous analysis. Upon determining the level of movement is below a threshold value, the system adjusts previous parameters used to drive one or more of the loudspeakers. By adjusting previous parameters instead of performing a complete recalibration, the system provides a more efficient technique for ensuring that the loudspeakers continue to produce accurate sound for the listener. 1. A method for adjusting drive parameters for one or more loudspeakers , comprising:driving, by an audio source, the one or more loudspeakers to output sound based on a first set of drive parameters;detecting a change to one loudspeaker of the one or more loudspeakers based on one or more sensors integrated within a cabinet of the one loudspeaker;adjusting, in response to the detected change, the first set of drive parameters to account for the change of the one loudspeaker, wherein the adjustment produces a second set of drive parameters; anddriving, by the audio source, the one or more loudspeakers to output sound based on the second set of drive parameters.2. The method of claim 1 , wherein the change is a movement of the one loudspeaker claim 1 , the method further comprising:determining if a level of movement of the one loudspeaker is above a threshold level, wherein the adjustment of the first drive parameters to produce the second drive parameters is performed in response to determining that the level of movement is ...

ACOUSTIC WAVE IMAGE GENERATING APPARATUS AND CONTROL METHOD THEREOF

There are provided an ultrasound image generating apparatus, which generates a high-quality ultrasound image even in a deep portion of a subject, and a control method thereof. In an ultrasound image (Img), for a portion (Ar) equal to or less than a depth threshold value (D), a real scanning line (L) obtained from an acoustic wave echo signal is used. In the ultrasound image (Img), for a portion (Ar) deeper than the depth threshold value (D), an interpolation scanning line (L) located between the real scanning lines (L) is generated from an acoustic wave echo signal having a positional deviation between a focusing position of ultrasound waves and an observation target position. Also for a portion deeper than the interpolation scanning line (L), a high-quality ultrasound image (Img) is obtained. 1. An acoustic wave image generating apparatus , comprising:an acoustic wave probe in which a plurality of acoustic wave transducers are arranged in an arc shape;a driving device for making the acoustic wave transducers transmit acoustic waves converging on a focusing position to a subject while updating the acoustic wave transducers for transmitting acoustic waves in a sequential manner;a real scanning line generation device for generating real scanning lines indicating an acoustic wave image of the subject using an acoustic wave echo signal indicating an acoustic wave echo from an observation target position of the subject that is obtained based on driving of the acoustic wave transducers by the driving device;an interpolation scanning line generation device for generating a first interpolation scanning line located between the real scanning lines using the acoustic wave echo signal, which has a positional deviation in the arc direction between the focusing position and the observation target position, for a portion deeper than a depth threshold value in the subject; andan acoustic wave image generation device for generating an acoustic wave image of the subject from the ...

METHOD, DEVICE, AND SYSTEM FOR NOISE REDUCTION IN MULTI-MICROPHONE ARRAY

To solve the problems with the prior art that a multi-microphone array cannot inhibit broad-band noises well and cannot be used in the increasingly widespread broad-band communication, embodiments of the present invention disclose a method, a device and a system for eliminating noises with multi-microphone array. The method according to an embodiment of the present invention comprises according to the number of different spacings between each of pairs of microphones of the multi-microphone array, dividing a full frequency band into the same number of sub-bands; decomposing signals of each of the pairs of microphones with the different spacings into a corresponding one of the sub-bands, wherein the larger the spacing between each pair of microphones is, the lower the frequencies of the sub-band into which the signals of the pair of microphones are decomposed will be; adaptively reducing the noises in the decomposed signals of each of the pairs of microphones with the different spacings in ...

REAL-TIME QUALITY MONITORING OF SPEECH AND AUDIO SIGNALS IN NOISY REVERBERANT ENVIRONMENTS FOR TELECONFERENCING SYSTEMS

Device for detecting sound sources and method for operating the same

Device for fixing sound sources The device has a carrier (1) with a number of microphones (2) mounted according to a scale with equal intervals, a recording device (3) connected to the microphone so as to transfer signals and tuned to record all signals from a microphone separately and an evaluation device for determining the position and acoustic power of the sound sources. A reduced number of microphones is arranged on the carrier according to a defined pattern whilst maintaining the scale separations to effect an essentially redundancy free storage of microphone signals in the recording device.

Selektive Schallaufnahmevorrichtung für reflektierende und geräuschvolle Umgebung

POSITION DIRECTED ACOUSTIC ARRAY AND BEAMFORMING METHODS

Verfahren und Systeme sind zum Empfangen erwünschter Töne vorgesehen. Das System enthält einen Positionssensor, der zum Bestimmen einer Insassenposition eines sprechenden Insassen innerhalb eines definierten Raumes und Übertragen der Position des sprechenden Insassen konfiguriert ist. Eine Vielzahl von Mikrofonen ist konfiguriert, um Schall aus dem Inneren des definierten Raumes zu empfangen und Audiosignale zu übertragen, die dem empfangenen Schall entsprechen. Ein Prozessor, der mit dem Positionssensor und den Mikrofonen in Verbindung steht, ist zum Empfangen der Position des sprechenden Insassen und der Audiosignale, Anwenden eines Beamformers auf die Audiosignale, um einen Mikrofonstrahl in Richtung der Insassenposition zu richten, und Erzeugen eines Strahlenformer-Ausgangssignals konfiguriert.

Wind noise rejection apparatus

Loudspeaker system with controlled directional sensitivity

LINEAR FILTERING FOR NOISE-SUPPRESSED SPEECH DETECTION

Systems and methods for suppressing noise and detecting voice input in a multi- channel audio signal captured by a plurality of microphones include (i) capturing a first audio signal via a first microphone and a second audio signal via a second microphone, wherein the first and second audio signals respectively comprises first and second noise content from a noise source; (ii) identifying the first noise content in the first audio signal; (iii) using the identified first noise content to determine an estimated noise content captured by the plurality of microphones; (iv) using the estimated noise content to suppress the first and second noise content in the first and second audio signals; (v) combining the suppressed first and second audio signals into a third audio signal; and (vi) determining that the third audio signal includes a voice input comprising a wake word.

METHOD AND APPARATUS FOR LOCALIZATION OF AN ACOUSTIC SOURCE

An acoustic signal processing method and system using a pair of spatially separated microphones (10, 11) to obtain the direction (80) or location of speech or other acoustic signals from a common sound source (2) are disclosed. The invention includes a method and apparatus for processing the acoustic signals by determining whether signals acquired during a particular time frame represent the onset (45) or beginning of a sequence of acoustic signals from the sound source, identifying acoustic received signals representative of the sequence of signals, and determining the direction (80) of the source based upon the acoustic received signals. The invention has applications to videoconferencing where it may be desirable to automatically adjust a video camera, such as by aiming the camera in the direction of a person who has begun to speak.

METHOD AND APPARATUS FOR LOCALIZATION OF AN ACOUSTIC SOURCE

An acoustic signal processing method and system using a pair of spatially separated microphones (10, 11) to obtain the direction (80) or location of speech or other acoustic signals from a common sound source (2) are disclosed. The invention includes a method and apparatus for processing the acoustic signals by determining whether signals acquired during a particular time frame represent the onset (45) or beginning of a sequence of acoustic signals from the sound source, identifying acoustic received signals representative of the sequence of signals, and determining the direction (80) of the source based upon the acoustic received signals. The invention has applications to videoconferencing where it may be desirable to automatically adjust a video camera, such as by aiming the camera in the direction of a person who has begun to speak.

Spiral-shaped array for broadband imaging

ACOUSTIC ANTENNA FOR DATA-PROCESSING WORK STATION

L'invention concerne une antenne acoustique pour station de travail informatique. Elle comprend une pluralité de microphones (Mi ) connectés à un circuit sommateur. Les microphones (Mi ) sont répartis en un agencement pour former une ligne sensiblement rectiligne et espacés chacun par rapport à un microphone de référence (Mir ) placé au voisinage de l'axe de symétrie vertical de l'écran de la station de travail selon une loi déterminée. L'agencement présente un diagramme de directivité sensiblement cylindrique dont l'axe de révolution est formé par la ligne rectiligne. Application à la réalisation de stations de travail informatiques, notamment pour visioconférence.

SYSTEMS, METHODS, APPARATUS, AND COMPUTER-READABLE MEDIA FOR ORIENTATION-SENSITIVE RECORDING CONTROL

Systems, methods, apparatus, and machine-readable media for orientation-sensitive selection and/or preservation of a recording direction using a multi-microphone setup are described.

REAL-TIME QUALITY MONITORING OF SPEECH AND AUDIO SIGNALS IN NOISY REVERBERANT ENVIRONMENTS FOR TELECONFERENCING SYSTEMS

A method for real-time monitoring of audio signals reception quality includes receiving output signals from a plurality of microphone clusters, each microphone cluster having at least two microphone units to receive audio signals from at least two distinct directions and output corresponding electrical signals; identifying comparative features of output signals for each of the microphone clusters; and selecting at least one microphone cluster based on the identified features. A system for real-time monitoring of audio signals reception quality includes a plurality of microphone clusters, each microphone cluster having at least two microphone units to receive audio signals from at least two distinct directions and output corresponding electrical signals; and a main audio unit to identify comparative features of output signals for each of the microphone clusters and to select at least one microphone cluster based on the identified features.

BEAM FORMING ARRAY OF TRANSDUCERS

A two-dimensional array of a plurality of transducers comprising a first plurality of like sub-arrays (11, 11a, 11b) of transducers (10) in a circularly symmetric arrangement around a common centre (C), where the transducers in each sub-array of the first plurality have individual distances from the common centre that form a progressive series of distances with a first lower limit and a first upper limit. Each sub-array in the first plurality of sub-arrays comprises at least three transducers arranged on a first straight line (12), and the first straight line is offset laterally a first distance (d) from the common centre. The number of sub-arrays is odd, and the sub-arrays may be separate units that can be selectively assembled to form the two-dimensional array and selectively disassembled.

Method and apparatus for steerable and endfire superdirective microphone arrays with reduced analog-to-digital converter and computational requirements

An end fire microphone array having reduced analog-to-digital converter requirements is disclosed. Analog filters are used to band-limit at least two secondary microphone elements which are spaced from a primary microphone element a distance respective of their band limited outputs. The band-limited secondary microphone outputs are combined by an analog summer and the primary microphone and combined secondary microphone signals are digitized by an analog-to-digital converter. A signal processor performs a super-directive analysis of the primary microphone signal and the combined secondary microphone signals. A steerable superdirective microphone array is disclosed. A plurality of microphones are arranged in a ring. The microphone outputs are digitized, split into frequency bands, and weighted sums are formed for each of a plurality of directions. A steering control circuit evaluates the relative energy of each directional signal in each band and selects a microphone direction for further ...

Spatial low-crosstalk headset

An apparatus for reducing cross-talk between transmitted audio signals and received audio in a headset. The headset includes one or more of a set of earphones, a headset frame, a microphone boom with an array of MEMS microphone configured to isolate the earphone audio from the microphone audio, a VOX circuit, low crosstalk cable(s), and/or other components. Sets of microphones may be enabled and/or disabled to reduce cross-talk between received audio signals and transmitted audio signals. The VOX circuit is configured to reduce cross-talk between received audio signals and transmitted audio signals.

Redundanzlose kreissymmetrische ebene Gruppenantenne

SCHALLSTRAHLUNGSMUSTERSTEUERUNG

Eine oder mehrere Ausführungsformen der vorliegenden Offenbarung benutzen zwei distinkt unterschiedliche Strahlungsvorrichtungen, die in dieselbe Richtung gerichtet sind, um ein abgeleitetes Schallstrahlungsmuster zu erstellen. Die Strahlungsvorrichtungen können sich in Bezug auf das Schallstrahlungsmuster, das jede Strahlungsvorrichtung individuell erzeugt, distinkt unterscheiden. Das abgeleitete Schallstrahlungsmuster ist einmalig für die individuellen akustischen Schallstrahlungsmuster einer der beiden Strahlungsvorrichtungen. Die Manipulation verschiedener Schlüsseldesignvariablen ermöglicht die Ableitung einer Vielzahl einmaliger Muster auf diese Art und Weise unter Verwendung nur der zwei Strahlungsvorrichtungen. Dies ermöglicht es wiederum, ein Schallstrahlungsmuster herzustellen, das zur einmaligen Geometrie eines Raumes passt.

Sound apparatus

Optimum broadcast audio capturing apparatus, method and system

The OPTIMUM BROADCAST AUDIO CAPTURING APPARATUS, METHOD AND SYSTEM (“OBAC”) transforms selection request, video feeds, and, audio feeds inputs via OBAC components into synchronized and optimum video and audio outputs. In one embodiment, a processor-implemented method for capturing optimal audio in sports broadcasting is disclosed, comprising: receiving real time data from a first instrumented camera situated in a venue; processing the real time data to determine a first field position in the venue; activating a first microphone from a plurality of microphones in a first microphone array based on the first field position; and sending a first audio selection from the first microphone.

Multi-microphone noise reduction using enhanced reference noise signal

Systems and methods of improved noise reduction include the steps of: receiving an audio signal from two or more acoustic sensors; applying a beamformer to employ a first noise cancellation algorithm; applying a noise reduction post-filter module to the audio signal including: estimating a current noise spectrum of the received audio signal after the application of the first noise cancellation algorithm, wherein the current noise spectrum is estimated using the audio signal received by the second acoustic sensor; determining a punished noise spectrum using the time-average level difference between the audio signal received by the first acoustic sensor and the current noise spectrum; determining a final noise estimate by subtracting the punished noise spectrum from the current noise spectrum; and applying a second noise reduction algorithm to the audio signal received by the first acoustic sensor using the final noise estimate; and outputting an audio stream with reduced background noise.

Methods, apparatuses and computer program products for facilitating directional audio capture with multiple microphones

An apparatus for providing directional audio capture may include a processor and memory storing executable computer program code that cause the apparatus to at least perform operations including assigning at least one beam direction, among a plurality of beam directions, in which to direct directionality of an output signal of one or more microphones. The computer program code may further cause the apparatus to divide microphone signals of the microphones into selected frequency subbands wherein an analysis performed. The computer program code may further cause the apparatus to select at least one set of microphones of the apparatus for selected frequency subbands. The computer program code may further cause the apparatus to optimize the assigned at least one beam direction by adjusting a beamformer parameter(s) based on the selected set of microphones and at least one of the selected frequency subbands. Corresponding methods and computer program products are also provided.

Conference System with a Microphone Array System and a Method of Speech Acquisition in a Conference System

A conference system is provided that includes a microphone array unit having a plurality of microphone capsules arranged in or on a board mountable on or in a ceiling of a conference room. The microphone array unit has a steerable beam and a maximum detection angle range. The conference system comprises a processing unit which is configured to receive the output signals of the microphone capsules and to steer the beam based on the received output signal of the microphone array unit. The processing unit is configured to control the microphone array to limit the detection angle range to exclude at least one predetermined exclusion sector in which a noise source is located. 1: A conference system , comprising:a microphone array having a plurality of microphone capsules arranged in or on a sound-reflecting board mountable on or in a ceiling of a conference room wherein the sound-reflecting board has an upper side that is directed to the ceiling when the microphone array is mounted on or in the ceiling of a conference room, wherein the sound-reflecting board has a lower side opposite the upper side, and wherein the microphone capsules are arranged in or on the lower side of the sound-reflecting board and adapted for acquiring sound coming from the conference room; anda processing unit configured to receive output signals of the microphone capsules and to execute audio beam forming based on the received output signals of the microphone capsules for predominantly acquiring sound coming from an audio source in a first direction; a delay control unit; and', 'a delay unit for each of the output signals of the microphone capsules, each delay unit configured to receive input from the delay control unit;, 'wherein the processing unit compriseswherein the delay control unit calculates individual delay values for each of the delay units according to a given direction.220-: (canceled)21: The conference system of claim 1 ,wherein the sound-reflecting board is a carrier board that is ...

Conference System with a Microphone Array System and a Method of Speech Acquisition in a Conference System

A conference system is provided that includes a microphone array unit having a plurality of microphone capsules arranged in or on a board mountable on or in a ceiling of a conference room. The microphone array unit has a steerable beam and a maximum detection angle range. The conference system comprises a processing unit which is configured to receive the output signals of the microphone capsules and to steer the beam based on the received output signal of the microphone array unit. The processing unit is configured to control the microphone array to limit the detection angle range to exclude at least one predetermined exclusion sector in which a noise source is located. 1: A conference system , comprising:a microphone array having a plurality of microphone capsules arranged in or on a board mountable on or in a ceiling of a conference room, wherein the microphone capsules are adapted for acquiring sound coming from the conference room; anda processing unit configured to receive output signals of the microphone capsules and to execute audio beam forming based on the received output signals of the microphone capsules for predominantly acquiring sound coming from an audio source in the conference room; a direction recognition unit configured to identify a direction of the audio source, wherein the direction recognition unit is configured to process the output signals of at least two of the microphone capsules, the processing comprising using a Steered Response Power with Phase Transform (SRP-PHAT) algorithm to calculate a score for each of a plurality of points in space that form a pre-defined search grid, and wherein the direction recognition unit outputs a direction signal indicating said direction of the audio source;', 'a delay control unit; and', 'a delay unit for each of the output signals of the microphone capsules, each delay unit configured to receive input from the delay control unit;', 'wherein the delay control unit calculates individual delay values for ...

MICROPHONE

A microphone comprises: a vibration member having a plate shape and formed so as to have elasticity and cause bending due to sound waves; a case having a depressed groove, which forms an air layer between the depressed groove and the vibration member since the vibration member is stacked on an upper part thereof, and coupled to the upper part of the depressed groove such that the vibration member is vibrated by sound waves; and a vibration detection unit provided on the vibration unit and the case and measuring the vibration of the vibration member vibrating by sound waves on the case, wherein when the vibration member vibrates, the air in the air layer flows so as to damp the vibration of the vibration member. 1. A microphone comprising:a vibration member having a plate shape and formed so as to have elasticity and cause bending due to sound waves;a case having a depressed groove, which forms an air layer between the depressed groove and the vibration member since the vibration member is stacked on an upper part thereof, and coupled to the upper part of the depressed groove such that the vibration member is vibrated by sound waves; anda vibration detection unit provided on the vibration unit and the case and measuring the vibration of the vibration member vibrating by sound waves on the case,wherein when the vibration member vibrates, the air in the air layer flows so as to damp the vibration of the vibration member.2. The microphone of claim 1 , wherein the case includes a coupling protrusion which is formed to protrude upward in the depressed groove and in which the vibration member is seated on an upper end claim 1 , andthe vibration member includes a coupling part coupled with the coupling protrusion while facing the coupling protrusion on the bottom so as to be vibrated by external force around the center in a horizontal direction.3. The microphone of claim 2 , wherein the coupling protrusion and the coupling part are integrally configured to have elasticity ...

Two-way loudspeaker with floating waveguide

One or more embodiments of the present disclosure relate to a two-way loudspeaker design that forces a condensed geometry between low frequency (LF) and high frequency (HF) drivers and then “floats” a midrange waveguide in front of the LF driver. This is a hybrid design meant to benefit from the close proximity of acoustic centers without introducing a central axis obstruction for the LF driver. In addition, the LF and HF waveguides and associated acoustic elements are used to redirect very low frequency energy not supported adequately by the LF waveguide to exit freely using other paths.

ACOUSTIC RADIATION PATTERN CONTROL

One or more embodiments of the present disclosure utilize two distinctly different radiation devices aimed in the same direction to create a derived acoustic radiation pattern. The radiation devices may be distinctly different in terms of the acoustic radiation pattern each radiation device generates individually. The derived acoustic radiation pattern is unique to the individual acoustic radiation patterns of either of the two radiation devices. Manipulation of several key design variables allows a multitude of unique patterns to be derived in this way using only the two radiation devices. In turn, this allows for engineering an acoustic radiation pattern to match the unique geometry of a room. 1. A dual-array loudspeaker comprising:a primary transducer producing a primary radiation pattern in a primary plane; anda secondary transducer positioned a distance in the primary plane from the primary transducer and producing a secondary radiation pattern different from the primary radiation pattern in the primary plane,wherein the secondary radiation pattern modifies the primary radiation pattern to produce a derived primary radiation pattern different from the primary and secondary radiation patterns in the primary plane.2. The dual-array loudspeaker of wherein the primary and secondary transducers include central axes of radiation claim 1 , wherein the central axes of radiation of the primary and secondary transducers lie on the primary plane.3. The dual-array loudspeaker of wherein the central axes of radiation of the primary and secondary transducers are generally parallel and spaced apart by the distance in the primary plane.4. The dual-array loudspeaker of wherein the secondary transducer manipulates the primary radiation pattern in the primary plane to achieve the derived primary radiation pattern oriented at an angle different than the central axes of radiation of the primary and secondary transducers.5. The dual-array loudspeaker of wherein the primary plane is ...

MICROPHONE ARRAYS

A system for capturing sound comprising a plurality of discrete microphones () and a processing system (). The plurality of discrete microphones are arranged in a circular array. The processing system () 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 () 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 () arranged in a circular array is also described. 1. A method comprising:receiving sound at a plurality of discrete microphones arranged in a circular array, at least some of said microphones producing signals in response to said sound,performing a first signal processing algorithm on sound originating from one or more of a first set of directions relative to said array to isolate a first sound source therein; andperforming a second signal processing algorithm on sound originating from one or more of a second set of directions relative to said array to isolate a second sound source therein.2. The method as claimed in wherein the first processing algorithm comprises a broadside processing technique.3. The method as claimed in wherein the first set of directions comprises directions up to a threshold angle from a perpendicular to the plane of the array.4. The method as claimed in wherein the threshold angle is between 50° and 70°.5. The method as claimed in wherein the second processing algorithm comprises super-directive beamforming.6. The method as claimed in further comprising using an orientation sensor to determine an orientation of the array and using said orientation to determine a first and second portions of sound.7. The method as claimed in comprising receiving sound at a plurality of discrete ...

Acoustic source separation

A system for directionally selective sound reception comprises an array of pressure sensors each arranged to output a pressure signal indicative of pressure, and processing means arranged to receive the pressure signals, identify a plurality of frequency components of the signals, identify at least one source direction, and identify at least one of the components as coming from the source direction. The sensor array comprises support means having two opposite sides and four sensors, at least one of the sensors being supported on each of the sides of the support means.

MACHINING SYSTEM AND MONITORING METHOD

The invention provides a machining system () comprising: a machining apparatus (), notably an abrasive waterjet cutting system (), said machining apparatus being adapted for machining a workpiece (); a monitoring device () adapted for monitoring machining conditions of the machining apparatus () and/or of the workpiece, the monitoring device comprising a plurality of sensors, said plurality of sensors comprising a first sensor () at a first location and a second sensor () at a second location which is distant from the first location. The plurality of sensors comprises a fourth sensor () which is formed by an array of microphones () arranged on a grid. The plurality of sensors comprises accelerometers, strain gauges and microphones. The invention also provides a monitoring method of a machining system () wherein a specific benchmark signature is chosen from a library. 1. A machining system comprising:a machining apparatus, comprising a cutting machine, said machining apparatus being adapted for machining a workpiece, the machining apparatus comprising a nozzle having an inlet end and an outlet end; 'a first sensor configured to be disposed at a first location downstream of the inlet end and a second sensor configured to be disposed at a second location downstream of the inlet end which is distant from the first location.', 'a monitoring device adapted for monitoring machining conditions of the machining apparatus, the monitoring device comprising a plurality of sensors, said plurality of sensors comprising2. The machining system of claim 1 , wherein the first sensor comprises a first strain gauge sensor claim 1 , and the second sensor comprises a second strain gauge sensor.3. The machining system in accordance with claim 2 , wherein the plurality of sensors further comprise a third strain gauge sensor at a third location which is distant from the first location and the second location claim 2 , and a fourth strain gauge sensor configured to be in contact of the ...

Orientation-Based Playback Device Microphone Selection

Aspects of a multi-orientation playback device including at least one microphone array are discussed. A method may include determining an orientation of the playback device which includes at least one microphone array and determining at least one microphone training response for the playback device from a plurality of microphone training responses based on the orientation of the playback device. The at least one microphone array can detect a sound input, and the location information of a source of the sound input can be determined based on the at least one microphone training response and the detected sound input. Based on the location information of the source, the directional focus of the at least one microphone array can be adjusted, and the sound input can be captured based on the adjusted directional focus. 1. A playback device comprising:a top face;a bottom face opposite the top face, wherein the bottom face comprises a longest face of the playback device;a side face perpendicular to the top face and the bottom face; a first subset of one or more microphones positioned along the top face; and', 'a second subset of one or more microphones positioned along the side face;, 'a set of microphones comprisingan orientation sensor;one or more processors; and determining that the playback device is in a horizontal orientation based on an indication from the orientation sensor that a longest face of the playback device is substantially parallel to a surface on which the playback device is positioned, wherein, in the horizontal orientation, the bottom face is substantially parallel to the surface and the side face is substantially perpendicular to the surface;', 'selecting the first subset of one or more microphones instead of the second subset of one or more microphones for sound input detection based on the determination that the playback device is in the horizontal orientation;', 'detecting, via the first subset of one or more microphones, a sound input;', 'processing ...

Sound Signal Processing Method and Apparatus

A sound signal processing method and apparatus are provided that relate to the audio signal processing field. The method in the present invention includes acquiring, by a mobile terminal, sound signals from a three-dimensional sound field, where at least three microphones are disposed on the mobile terminal and one microphone is configured to receive a sound signal in at least one direction; acquiring, according to the acquired sound signals, a direction of a sound source relative to the mobile terminal; and obtaining spatial audio signals according to the direction of the sound source relative to the mobile terminal and the acquired sound signals, where the spatial audio signals are used for simulating the three-dimensional sound field. The present invention is applicable to a process of collecting and processing signals in a three-dimensional sound field surrounding a terminal.

Linear Filtering for Noise-Suppressed Speech Detection

Systems and methods for suppressing noise and detecting voice input in a multi-channel audio signal captured by a plurality of microphones include (i) capturing a first audio signal via a first microphone and a second audio signal via a second microphone, wherein the first and second audio signals respectively comprises first and second noise content from a noise source; (ii) identifying the first noise content in the first audio signal; (iii) using the identified first noise content to determine an estimated noise content captured by the plurality of microphones; (iv) using the estimated noise content to suppress the first and second noise content in the first and second audio signals; (v) combining the suppressed first and second audio signals into a third audio signal; and (vi) determining that the third audio signal includes a voice input comprising a wake word.

LOUDSPEAKER SYSTEM AUDIO RECOVERY IMAGING AMPLIFIER

The present invention is a small high fidelity full spectrum speaker, which arranges a passive speaker and two active speakers such that fidelity of a larger speaker or one with a cross-over network is achieved. 1. A full spectrum speaker system without use of crossover networks comprising:a) a speaker enclosure having an inside and a bottom;b) a bass chamber located at the bottom inside of the speaker enclosure ported outside the enclosure;c) a full spectrum baffle mounted passive speaker ported inside the enclosure and positioned directly above and facing the ported bass;d) a first full spectrum baffle mounted active speaker ported inside the enclosure, positioned directly above the passive speaker and facing downward toward the back of the passive speaker in the same direction as the passive speaker; ande) a second full spectrum baffle mounted active speaker mounted ported outside of the enclosure and in a different direction than the first speaker, the second full spectrum speaker being wired out of phase with the first speaker.2. The speaker system according to which further comprises at least two tubular cylinders positioned inside the enclosure.3. The speaker system according to which comprises a first tubular cylinder positioned inside the enclosure separating the first speaker and the second speaker.4. The speaker system according to which comprises a second tubular cylinder separating the passive speaker and the first speaker.5. The speaker system according to wherein a diaphragm of the first speaker and the passive speaker are aligned along the axis of their diaphragm.6. The speaker system according to wherein the direction that the second speaker is ported claim 1 , is adjustable.7. The speaker system according to wherein the second speaker is positioned about 135 degrees different than the position of the first speaker.8. The speaker system according to which further comprises at least one full spectrum baffle mounted passive speaker ported outside the ...

METHODS, APPARATUSES AND COMPUTER PROGRAM PRODUCTS FOR FACILITATING DIRECTIONAL AUDIO CAPTURE WITH MULTIPLE MICROPHONES

An apparatus for providing directional audio capture may include a processor and memory storing executable computer program code that cause the apparatus to at least perform operations including assigning at least one beam direction, among a plurality of beam directions, in which to direct directionality of an output signal of one or more microphones. The computer program code may further cause the apparatus to divide microphone signals of the microphones into selected frequency subbands wherein an analysis performed. The computer program code may further cause the apparatus to select at least one set of microphones of the apparatus for selected frequency subbands. The computer program code may further cause the apparatus to optimize the assigned at least one beam direction by adjusting a beamformer parameter(s) based on the selected set of microphones and at least one of the selected frequency subbands. Corresponding methods and computer program products are also provided. 1. An apparatus comprising:a directional audio capture module; anda plurality of microphones positioned at predetermined locations of the apparatus; assign at least one beam direction, among a plurality of beam directions, in which to direct directionality of an output signal of one or more of the microphones;', 'select a first set of microphones comprising at least two microphones, of the plurality of microphones, for a first frequency band;', 'select a second set of microphones comprising at least two microphones, of the plurality of microphones, for a second frequency band; and', 'optimize the assigned at least one beam direction by adjusting at least one beamformer parameter based on the selected first and second sets of microphones., 'the directional audio capture module is configured to2. The apparatus of claim 1 , wherein the directional audio capture module is further configured to:select the first set of microphones and the second set of microphones based in part on a distance between ...

DIRECTIONAL AUDIO CAPTURE

Systems and methods for improving performance of a directional audio capture system are provided. An example method includes correlating phase plots of at least two audio inputs, with the audio inputs being captured by at least two microphones. The method can further include generating, based on the correlation, estimates of salience at different directional angles to localize a direction of a source of sound. The method can allow providing cues to the directional audio capture system based on the estimates. The cues include attenuation levels. A rate of change of the levels of attenuation is controlled by attack and release time constants to avoid sound artifacts. The method also includes determining a mode based on an absence or presence of one or more peaks in the estimates of salience. The method also provides for configuring the directional audio capture system based on the determined mode. 1. A method for improving performance of a directional audio capture system , the method comprising:correlating phase plots of at least two audio inputs;generating, based on the correlation, estimates of salience at different directional angles to localize at least one direction associated with at least one source of a sound;determining cues based on the estimates of salience; andproviding the cues to the directional audio capture system.2. The method of claim 1 , wherein the cues are used by the directional audio capture system to attenuate or amplify the at least two audio inputs at the different directional angles.3. The method of claim 1 , wherein the cues include at least attenuation levels of the different directional angles.4. The method of claim 1 , wherein the estimates of salience include a vector of saliences at directional angles from 0 to 360 in a plane parallel to a ground.5. The method of claim 1 , wherein generating the cues includes mapping the different directional angles to relative levels of attenuation for the directional audio capture system.6. The ...

Coprime microphone array system

A coprime microphone array (CMA) system, comprising: a CMA arrangement that includes a pair of uniform linear microphone subarrays that are coincident and have a coprime number of microphones; a computing system that processing signals from each microphone in the CMA arrangement and generates sound localization information.

Flexible spatial audio capture apparatus

An apparatus comprising: at least one determiner configured to determine at least one characteristic associated with a flexible part of a further apparatus; a signal processor configured to process at least one signal dependent on the at least one characteristic associated with the flexible part of the further apparatus, wherein the signal is at least one of an audio and a video signal; and an user interface generator configured to generate at least one user interface indication dependent on the characteristic associated with a flexible part of the further apparatus.

Distributed microphone array and sound source positioning system applicable thereto

Disclosed is a distributed microphone array. The array comprises: a plurality of distributed microphone array nodes, wherein each microphone array node is provided with one or a plurality of microphone audio acquisition modules; and the microphone array node is connected to an external signal processing unit by means of wireless network communication, and uploads an unprocessed audio signal or an audio signal which has undergone simple signal processing on the array node to the signal processing unit. In the present invention, all the wireless microphone array nodes distributed in a wireless network form a large microphone array. In a wireless microphone array, microphone signal acquisition and transmission are based on a wireless type, and the distribution of microphones is no longer limited by a physical dimension or a cabling connection, so that the distribution of a significant amount of microphones can be achieved, and the distance between the microphones and the coverage area of the microphone array can be greatly increased.

DIRECTIONAL AUDIO PICKUP IN COLLABORATION ENDPOINTS

A microphone array includes one or more front-facing microphones disposed on a front surface of the collaboration endpoint and a plurality of secondary microphones disposed on a second surface of the collaboration endpoint. The sound signals received at each of the one or more front-facing microphones and the plurality of secondary microphones are converted into microphone signals. When the sound signals have a frequency below a threshold frequency, an output signal is generated from microphone signals generated by the one or more front-facing microphones and the plurality of secondary microphones. When the sound signals have a frequency at or above a threshold frequency, an output signal is generated from microphone signals generated by only the one or more front-facing microphones. 1. A method comprising:receiving, with a microphone array of an apparatus, sound signals comprising a plurality of frequency components, wherein the microphone array includes one or more front-facing microphones disposed on a front surface of the apparatus and one or more secondary microphones disposed on a second surface of the apparatus;converting frequency components of the sound signals received at each of the one or more front-facing microphones and the one or more secondary microphones into microphone signals;for frequency components of the sound signals having a frequency below a threshold frequency, generating output signals from microphone signals generated by the one or more front-facing microphones and from microphone signals generated by the one or more secondary microphones; andfor frequency components of the sound signals having a frequency at or above the threshold frequency, generating output signals from only the microphone signals generated by one or more front-facing microphones.2. The method of claim 1 , wherein the front surface of the apparatus is substantially orthogonal to the second surface of the apparatus.3. The method of claim 1 , wherein the one or more ...

SPEAKER MODULE, ELECTRONIC DEVICE INCLUDING THE SPEAKER MODULE, AND DISPLAY DEVICE INCLUDING THE ELECTRONIC DEVICE

A speaker module, An electronic device, and a display device include a housing, a main board, and at least one speaker module placed in the housing. The at least one speaker module includes at least one first speaker unit disposed in a first direction and at least one second speaker unit disposed in a second direction different from the first direction and having at least one side of the at least one second speaker unit contacting with at least one side of the at least one first speaker unit. Driving vibrations may be offsetted because the speaker units are disposed in different directions. 1. An electronic device , comprising:a housing providing an internal space;a shield rib installed inside the housing and dividing the internal space into a upper space and a lower space,a first speaker unit installed inside the upper space, dividing the upper space into a first front space and a first rear space, and disposed in a first direction; anda second speaker unit installed inside the lower space, dividing the lower space into a second front space and a second rear space, and disposed in a second direction,wherein the shield rib comprises:a first through hole connecting the first front space to the second rear space; anda second through hole connecting the first rear space to the second first space.wherein the housing comprises:a first opening formed on the housing, and connecting the first front space and an outside of the housing; anda second opening formed on the housing, and connecting the first rear space and the outside of the housing.2. The electronic device of claim 1 , wherein the first speaker unit comprises a first magnet placed at a back side of the first speaker unit and a first cone paper placed at a front side of the first speaker unit claim 1 , and the second speaker unit comprises a second magnet placed at a back side of the second speaker unit and a second cone paper placed at a front side of the second speaker unit claim 1 ,a back surface of the first ...

VARIABLE DEVICE FOR DIRECTING SOUND WAVEFRONTS

A variable device for directing sound wavefronts. In one example, individual housings including sound transducers can be shifted relative to one another in such a way that a wavefront adapted to the respective reproduction requirements can be produced. This configuration can be re-designed by mechanically shifting the individual elements of the variable device again. 1. A variable device for directing sound wavefronts , comprising:more than two individual housings each of which comprises a sound transducer, wherein the individual housings can be shifted relative to one another by means of a positive connection between the individual housings, in such a way that the acoustic centres of the sound transducers can be directed in two spatial axes with respect to the shape of the sound wavefront to be generated such that, depending on the displacement of the housings relative to one another, both concave and convex sound wavefronts can be generated by the device if all transducers are supplied with the same audio signal from a common source.2. The variable device for directing sound wavefronts according to claim 1 , whereinthe acoustic centres of the sound transducers can be arranged on a surface with a radius of curvature, wherein the surface can be defined by the position of a virtual origin of the sound wavefront which is located at the centre of the radius of curvature, and from which the direction and opening angle of the radiated sound wavefront can be determined if the virtual origin is positioned behind the device, and at which the elementary waves of the individual sound transducers sum together in phase, so that a focused sound source is produced when it is positioned in front of the device.3. The variable device for directing sound wavefronts according to claim 1 , whereinthe diameter of the individual sound transducers is smaller than a wavelength of a fundamental tone of the audio signal.4. The variable device for directing sound wavefronts according to claim ...

FLEXIBLE GEOGRAPHICALLY-DISTRIBUTED DIFFERENTIAL MICROPHONE ARRAY AND ASSOCIATED BEAMFORMER

A differential microphone array includes a plurality of microphones situated on a substantially planar platform and a processing device, communicatively coupled to the plurality of microphones, to receive a plurality of electronic signals generated by the plurality of microphones responsive to a sound source and execute a minimum-norm beamformer to calculate an estimate of the sound source based on the plurality of electronic signals, wherein the minimum-norm beamformer is determined subject to a constraint that an approximation of a beampattern associated with the differential microphone array substantially matches a target beampattern. 1. A differential microphone array comprising:a plurality of microphones located on a substantially planar platform; and receive a plurality of electronic signals generated by the plurality of microphones responsive to a sound source; and', 'execute a minimum-norm beamformer to calculate an estimate of the sound source based on the plurality of electronic signals, wherein the minimum-norm beamformer is determined subject to a constraint that an approximation of a beampattern associated with the differential microphone array substantially matches a target beampattern., 'a processing device, communicatively coupled to the plurality of microphones, to2. The differential microphone array of claim 1 , wherein each one of the plurality of electronic signals represents a respective version of the sound source received at a corresponding one of the plurality of microphones.3. The differential microphone array of claim 1 , further comprising:an analog-to-digital converter, communicatively coupled to the plurality of microphones and the processing device, to convert the plurality of electronic signals into a plurality of digital signals.4. The differential microphone array of claim 1 , wherein the plurality of microphones are geographically-distributed at locations specified with respect to a reference point in a coordinate system on the ...

DIRECTIONAL AUDIO RECORDING SYSTEM

A directional audio recording system functions to allow certain audio information to be captured and recorded for later consumption. The selection of audio information for capture may be accomplished by ascertaining the direction of an audio source from a directionally discriminating acoustic sensor and isolating acoustic information originating from the direction so determined. Directional cues may also be recorded and a playback system may apply the directional cues to the stored information representing audio in a spatialization engine such as head-related transfer functions. 1. An directional recording system comprising:a directionally discriminating acoustic sensor;a beam forming unit connected to said directionally discriminating acoustic sensor;a location processor connected to said beam forming unit;a beam steering unit connected to said location processor and to said directionally discriminating acoustic sensor; anda digital storage unit connected to said beam steering unit.2. A directional recording system according to further comprising a record/playback controller connected to said digital storage unit.3. A directional recording system according to wherein said digital storage unit is connected to said location processor.4. A directional recording system according to wherein said digital storage unit stores information representing directionally isolated acoustic information.5. A directional recording system according to wherein said digital storage unit stores information representing directional cues corresponding to said directionally isolated acoustic information.6. A directional recording system according to further comprising a motion sensor connected to said location processor.7. A directional recording system according to wherein said location processor further comprises an accelerometer.8. A directional recording system according to wherein said directionally discriminating acoustic sensor is a microphone array.9. A directional recording system ...

PORTABLE MICROPHONE ARRAY

A microphone array of three or more microphones may be mounted on a housing or substrate configured to be part of a smartphone or a smartphone protective case. The microphone array may be positioned so that the far field sensing range of the microphone array is unobstructed. The microphone array may be utilized with a beam-forming system in order to determine location of an audio source and a beam-steering system in order to isolate audio emanating from the direction of the audio source. The beam-forming system may be suitable for tracking the movement of one or more audio sources in order to inform the beam-steering system of the direction or location to be isolated. 1. A microphone array comprising:a base associated with a mobile computing device;three or more microphones mounted on said base; wherein said microphones are mounted in a configuration with a first microphone mounted in a position that is not co-linear with a second microphone and a third microphone; anda fourth microphone mounted in a location that is not co-planar with said first microphone, said second microphone and said third microphone.2. A microphone array according to wherein said microphones are mounted on said base in a configuration where claim 1 , for every angle of azimuth referenced from said microphone array from 0 degrees to 360 degrees claim 1 , there are at least two microphones in said array which include the angle of azimuth within their field of sensitivity and are unobstructed by said base and user.3. A microphone array according to wherein said base is an outer housing of said mobile computing device.4. A microphone array according to wherein said base is a protective case configured to receive a mobile computing device.5. A microphone array according to further comprising an auxiliary power supply mounted in said protective case.6. A microphone array according to further comprising a detachable module mating with a housing of said protective case and wherein said microphones ...

Microphone Array System

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

REDUCING MULTIPLE SOURCES OF SIDE INTERFERENCE WITH ADAPTIVE MICROPHONE ARRAYS

Configuring microphones in an array to gather signals from a desired source of sound in a main lobe, and configuring the microphones to gather side interference by controlling at least one of a relative phase or a relative magnitude of signals gathered by each of the microphones to provide a respective plurality of differential microphone arrays each having a corresponding directional pattern that orients a null towards the desired source of sound and that provides a side lobe. The gathered side interference from the plurality of differential microphone arrays is processed to generate a reference signal that is subtracted from the signals that are gathered from the main lobe to provide an output signal wherein the side interference is reduced or cancelled. 1. A method comprising:configuring a plurality of microphones in an array to gather signals from a desired source of sound in a main lobe of the array;configuring the plurality of microphones to gather side interference by controlling at least one of a relative phase or a relative magnitude of signals gathered by each of the plurality of microphones to provide a respective plurality of microphone arrays each having a corresponding directional pattern that orients a null towards the desired source of sound and that provides one or more side lobes; andprocessing the gathered side interference from the plurality of microphone arrays to generate a reference signal that is subtracted from the signals that are gathered from the main lobe of the array to provide an output signal wherein the side interference is reduced or cancelled.2. The method of wherein the one or more side lobes comprise a first side lobe oriented for gathering side interference from a right side of the array and a second side lobe oriented for gathering side interference from a left side of the array.3. The method of wherein a first source of side interference is situated in the first side lobe and a second source of side interference is situated in ...

Microphone system

A microphone array includes an n microphone units (where n is an integer equal to or larger than three). A first one of the microphone units includes three microphones (( 14 - 1 through 14 - 3 )) arranged in line at equal intervals. An m-th microphone unit (where m is a positive integer expressed by 1<m<n) includes the microphones at either end of an (m−1)-th microphone unit, and another microphone spaced from the microphone at one end of the (m−1)-th microphone unit by a distance substantially equal to the distance between the microphones at either end of the (m−1)-th microphone unit, and disposed at a location on the side of the (m−1)-th microphone unit opposite to the side where the microphone at the one end at the one end of the (m−)-th microphone unit is disposed. The (m+1)-th microphone unit includes the microphones at either end of an m-th microphone unit, and another microphone spaced from the microphone at one end of the m-th microphone unit by a distance substantially equal to the distance between the microphones at either end of the m-th microphone unit, and disposed at a location on the side of the (m+1)-th microphone unit opposite to the side where the microphone at the one end of the (m+1)-th microphone unit is disposed.

Audio Signal Processing Method and Apparatus and Differential Beamforming Method and Apparatus

An audio signal processing method and apparatus and a differential beamforming method and apparatus to resolve a problem that an existing audio signal processing system cannot process audio signals in multiple application scenarios at the same time. The method includes determining a super-directional differential beamforming weighting coefficient, acquiring an audio input signal and determining a current application scenario and an audio output signal, acquiring, a weighting coefficient corresponding to the current application scenario, performing super-directional differential beamforming processing on the audio input signal using the acquired weighting coefficient in order to obtain a super-directional differential beamforming signal in the current application scenario, and performing processing on the formed signal to obtain a final audio signal required by the current application scenario. By using this method, a requirement that different application scenarios require different audio signal processing manners can be met.

DISTRIBUTED MICROPHONES SIGNAL SERVER AND MOBILE TERMINAL

A distributed microphones signal server includes a network interface circuit that communicates with mobile terminals and a plurality of microphones that are geographically distributed. The server receives an audio request message containing a location of the mobile terminal, and selects a microphone from among the plurality of microphones responsive to the location of the mobile terminal. The server obtains an audio stream from the selected microphone, and communicates the audio stream to the mobile terminal. Related methods by a distributed microphones signal server are disclosed. Related computer program products and methods by a mobile terminal are disclosed. 1. A distributed microphones signal server comprising:a network interface circuit configured to communicate with mobile terminals and a plurality of microphones that are geographically distributed;a processor connected to the network interface circuit; and receiving an audio request message from a mobile terminal which contains a location of the mobile terminal;', 'selecting a microphone from among the plurality of microphones responsive to the location of the mobile terminal;', 'obtaining an audio stream from the microphone that is selected; and', 'communicating the audio stream to the mobile terminal., 'a memory storing program code that is executed by the processor to perform operations comprising2. The distributed microphones signal server of claim 1 , wherein the operation to select a microphone from among the plurality of microphones responsive to the location of the mobile terminal claim 1 , comprises:determining a directional heading of the mobile terminal, and a listen-ahead distance responsive to the audio request message;determining a listen-ahead area responsive to a combination of the location of the mobile terminal, the directional heading of the mobile terminal, and the listen-ahead distance; andselecting the microphone from among the plurality of microphones responsive to proximity of a ...

OUTERWEAR-MOUNTED MULTI-DIRECTIONAL SENSOR

A multi-directional sensor may include a microphone array of three or more microphones mounted on outerwear. The microphone array may be positioned and configured so that its far field azimuth sensing range is unobstructed by the outerwear. An accelerometer may be provided and mounted in a location which is fixed with respect to the microphones of the microphone array. A beacon, such as an ultrasonic transmitter or BLE (Bluetooth Low Energy) transmitter may be associated with or attached to the outerwear. The microphone array may be utilized with a beam-forming system in order to determine location of an audio source and a beam-steering system in order to isolate audio emanating from the direction of the audio source. The beam-forming system is suitable for tracking the movement of the audio source in order to inform the beam-steering system of the direction or location to be isolated. Because the microphone array will move with a user, a motion sensor may be provided to reduce the computational resources required for tracking and isolation by allowing compensation for change in position and orientation of the user. The beacon will facilitate location of the wearer. 1. A body-mounted multi-directional sensor comprising:a base configured to be worn by a user;a multi-directional acoustic sensor mounted on said base and wherein said base is outerwear.2. A body-mounted multi-directional sensor wherein said multi-directional acoustic sensor further comprised three or more microphones mounted in a configuration with a first microphone mounted in a position that is not co-linear with a second microphone and a third microphone.3. A body-mounted multi-directional sensor according to further comprising a fourth microphone mounted in a location that is not co-planar with said first microphone claim 2 , said second microphone and said third microphone.4. A multi-directional sensor according to wherein said microphones are mounted on said base in a configuration where claim 3 , ...

SURROUND SOUND RECORDING ARRAY

Provided herein is a multichannel sound recording array having an elongated body and a plurality of arms provided at fixed angles relative to a center position of a recording field to enhance sound localization in a surround sound recording. 1. A multichannel sound recording array comprising:(a) an elongated body having an axis;(b) a sleeve slidingly disposed on the body and having a first surface, a second surface, and an outer-circumferential surface, wherein the first surface comprises a plurality of slots extending away from the axis at fixed angles relative to a center position of a recording field;(c) a plurality of arms, each having a first end hingedly attached to one of the plurality of slots of the sleeve and extending away from the axis through one of a plurality of slits disposed in the outer-circumferential surface of the sleeve;(d) a plurality of support bars, each having a proximal end hingedly attached to the body and a distal end slidingly attached to one of the plurality of arms; and(e) a plurality of microphones, each disposed on a second end of one of the plurality of arms,wherein when the sleeve is moved along the axis of the body in a direction towards the proximal ends of the plurality of support bars, the arms extend away from the axis of the body, thereby extending the microphones at fixed angles relative to the center position of the recording field, and wherein when the sleeve is moved along the axis of the body in a direction away from the distal ends of the plurality of supports, the arms retract towards the axis of the body.2. The multichannel sound recording array of claim 1 , wherein the multichannel sound recording array comprises four arms claim 1 , four support bars claim 1 , and the sleeve includes four slots.3. The multichannel sound recording array of claim 2 , wherein two of the slots are provided at 30° relative to the center position of the recording field claim 2 , and the other two slots are provided at 110° relative to the ...

ARRAYING SPEAKERS FOR A UNIFORM DRIVER FIELD

A method and system for a noise cancellation comprises an amplifier in communication with the three or more speakers disposed in an area. A system controller produces a driver signal for each of the speakers in response to a signal from at least one microphone detecting sound in the area and communicates the driver signals to the amplifier. The amplifier drives each speaker with the driver signal produced for that speaker. In response to the driver signals, the speakers emit sound that combined produces a substantially uniform sound pressure field for a particular zone within the area. The substantially uniform sound pressure field produced by the speakers has a magnitude and phase adapted to attenuate a noise field in the area corresponding to the sound detected by the at least one microphone. 1. A method of attenuating noise comprising:producing a driver signal for each of three or more speakers disposed in an area in response to a signal produced in response to sound detected within the area by at least one microphone; andgenerating within a particular zone in the area, by combined sound emitted by the three or more speakers in response to the driver signals, a substantially uniform sound pressure field that attenuates a noise field corresponding to the sound detected by the at least one microphone.2. The method of claim 1 , wherein the substantially uniform sound pressure field has a magnitude and phase that is adapted to attenuate the noise field corresponding to the sound detected by the at least one microphone.3. The method of claim 1 , further comprising arranging the three or more speakers along a common plane.4. The method of claim 1 , wherein the three or more speakers include a left speaker claim 1 , a center speaker claim 1 , and a right speaker claim 1 , the particular zone surrounds an expected location of a head of an occupant of the area claim 1 , the left and right speakers are disposed an equal distance from the expected location of the head of ...

DEVICE AND METHOD FOR DRIVING A SOUND SYSTEM AND SOUND SYSTEM

A driving device for a sound system by loudspeaker signals, wherein the sound system has a wave field synthesis loudspeaker array and one or several supply loudspeakers arranged separate from the wave field synthesis array includes an audio input for receiving at least one audio signal from at least one sound source, a position input for receiving information on a position of the sound source, a wave field synthesis unit for calculating loudspeaker signals for the loudspeakers of the wave field synthesis loudspeaker array, and a provider for providing the loudspeaker signal for the one or the several supply loudspeakers. The driving device enables a sound system by means of which sound localization becomes possible for the audience and at the same time pleasant levels can be achieved also in the first rows of the audience. 1. A method for driving a sound system by loudspeaker signals , the sound system comprising a wave field synthesis loudspeaker array and one or several supply loudspeakers arranged separate from the wave field synthesis loudspeaker array , the method comprising:receiving at least one audio signal from at least one sound source;receiving information on a position of a sound source of the at least one sound source;providing one or more first loudspeaker signals for supply loudspeakers of the one or several supply loudspeakers based on an audio signal of the at least one audio signal,calculating second loudspeaker signals for array loudspeakers of the wave field synthesis loudspeaker array based on the position of the sound source, based on the audio signal and based on a position of the array loudspeakers so that a sound field produced by the array loudspeakers allows localizing the sound source by a listener in a space supplied by the supply loudspeakers; andwherein the array loudspeakers and the supply loudspeakers are driven such that a wave front of the array loudspeakers arrives at a listener before a wave front of the supply loudspeakers.2. ...

Sound source localization device, sound processing system, and control method of sound source localization device

A sound source localization device, which has a plurality of sound pickup devices which record a sound signal and specifies a direction of a sound source based on sound signals recorded by at least two sound pickup devices of the sound pickup devices, includes a notification device that notifies information based on an arrangement of the sound pickup devices.

Method and Apparatus for Determining the Direction of Arrival of a Sonic Boom

A method and apparatus for determining a direction of arrival of an acoustic event of interest that arrives at an array. Data acquired from the array is received. A portion of the data that represents an acoustic signature for the acoustic event of interest is identified. An azimuth angle may be computed for an acoustic event using the portion of the data identified. An elevation angle for the acoustic event is computed using a beamforming process customized based on the azimuth angle computed.

SOUND REFLECTIONS FOR PORTABLE ASSEMBLIES

A sound reflection assembly comprises a rigid structure having a sound reflection section and a support section, a rigid part in the support section, etc. A directional sound generation module is placed on the rigid part. The directional sound generation module is spatially oriented to direct sound waves generated by one or more speaker elements in the directional sound generation module towards the sound reflection section to be reflected by the sound reflection section towards an audience in an audience area in front of the sound reflection assembly. The directional sound generation module may be a sound bar. 1. A sound reflection assembly , comprising:a rigid structure having a sound reflection section and a support section;a rigid part in the support section;a directional sound generation module placed on the rigid part, wherein the directional sound generation module is spatially oriented to direct sound waves generated by one or more speaker elements in the directional sound generation module towards the sound reflection section to be reflected by the sound reflection section towards an audience in an audience area in front of the sound reflection assembly.2. The sound reflection assembly as recited in claim 1 , wherein the support section represents a lower section of the rigid structure claim 1 , and wherein the sound reflection section is an upper section of the rigid structure.3. The sound reflection assembly as recited in claim 1 , wherein a sound reflective surface of the sound reflection section forms a contiguous surface with a front surface of the support section.4. The sound reflection assembly as recited in claim 1 , wherein the directional sound generation module is configured to direct the sound waves away from the support section.5. The sound reflection assembly as recited in claim 1 , wherein the support section comprises a vertical planar surface facing the audience.6. The sound reflection assembly as recited in claim 1 , wherein the sound ...

METHODS, APPARATUSES AND COMPUTER PROGRAM PRODUCTS FOR FACILITATING DIRECTIONAL AUDIO CAPTURE WITH MULTIPLE MICROPHONES

An apparatus for providing directional audio capture may include a processor and memory storing executable computer program code that cause the apparatus to at least perform operations including assigning at least one beam direction, among a plurality of beam directions, in which to direct directionality of an output signal of one or more microphones. The computer program code may further cause the apparatus to divide microphone signals of the microphones into selected frequency subbands wherein an analysis performed. The computer program code may further cause the apparatus to select at least one set of microphones of the apparatus for selected frequency subbands. The computer program code may further cause the apparatus to optimize the assigned at least one beam direction by adjusting a beamformer parameter(s) based on the selected set of microphones and at least one of the selected frequency subbands. Corresponding methods and computer program products are also provided. 1. An apparatus comprising:a directional audio capture module; anda plurality of microphones positioned at predetermined locations of the apparatus; assign at least one beam direction, among a plurality of beam directions, in which to direct directionality of an output signal of one or more of the microphones based on a respective selected application, among a plurality of applications;', 'select at least one of a first set of microphones comprising at least two microphones for a first frequency band and a second set of microphones comprising at least two other microphones for a second frequency band based in part on the selected application; and', 'optimize the output signal for the assigned at least one beam direction by adjusting at least one beamformer parameter based on the selected first and second sets of microphones, wherein respective sets of microphone signals for the first and second frequency bands are processed by the adjusted at least one beamformer parameter., 'the directional ...

Microphone Array System

A microphone array system or microphone array unit for a conference system is provided that includes a front board, side walls and a plurality of microphone capsules arranged in or on the front board mountable on or in a ceiling of a conference room. The microphone array system or unit is adapted for generating a steerable beam within a maximum detection angle range. The microphone array system or microphone array unit includes a processing unit which is configured to receive the output signals of the microphone capsules and to steer the beam based on the received output signal of the microphone array. The processing unit is configured to control the microphone array to limit the detection angle range to exclude at least one predetermined exclusion sector in which a noise source is located. 1. A microphone array system mountable on or in a ceiling of a conference room , the microphone array system comprising:a chassis comprising a front board and side walls, the front board comprising an upper side and a lower side and the side walls extending on the upper side of the front board;a plurality of microphone capsules arranged inside the chassis, wherein the microphone capsules are adapted for acquiring sound coming from below the microphone array system when mounted on or in a ceiling of a conference room; anda processing unit configured to receive output signals of the microphone capsules and to execute audio beam forming based on the received output signals of the microphone capsules for predominantly acquiring sound coming from an audio source in a first direction;wherein the microphone array system comprises or is mounted to a laterally protruding support frame;wherein the microphone array system is adapted for being mounted in a drop ceiling that is fixed to a ceiling of a conference room by support beams; andwherein the laterally protruding support frame is adapted for resting on said support beams such that the side walls of the microphone array system partially ...

Spatial Encoding Directional Microphone Array

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. 1. An article of manufacture comprising:a device body having a non-spheroidal shape;a plurality of microphones configured at a plurality of different locations on the device body, each microphone configured to generate a corresponding microphone signal from an incoming acoustic signal; anda signal processing system configured to process the microphone signals to generate a plurality of different output beampatterns in at least two non-parallel directions wherein the signal processing system is configured to generate at least one of the output beampatterns based on effects of the device body on the incoming acoustic signal.2. The article of claim 1 , wherein the device body has a general parallelepiped shape.3. The article of claim 1 , wherein the signal processing system comprises claim 1 , for the at least one output beampattern claim 1 , a signal processing subsystem comprising:a first diffraction filter configured to filter a first microphone signal to generate a first diffraction-filtered microphone signal, wherein the first diffraction filter is configured based on the effects of the device body on the incoming acoustic signal;a second diffraction filter different from the first diffraction filter and configured to filter a second microphone signal to generate a second ...

DIRECTION DETECTION DEVICE FOR ACQUIRING AND PROCESSING AUDIBLE INPUT

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. 1. A direction detection device configured to determine a direction from which an audible signal is received , comprising: determining a first time difference between when a first microphone received a first portion of an audible signal and when a second microphone received the first portion of the audible signal, and', 'determining a second time difference between when the first microphone received the first portion of the audible signal and when a third microphone received the first portion of the audible signal; and, 'a delay determination algorithm stored in a memory of an electronic device, wherein the delay determination algorithm comprises a number of instructions which, when executed by a processor, causes the electronic device to perform operations comprising 'determine a direction from which the audible signal was received based on a comparison of the first time difference and the second time difference with a plurality of time delay values that are stored in ...

DIRECTION DETECTION DEVICE FOR ACQUIRING AND PROCESSING AUDIBLE INPUT

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. 1. A method of processing an audible signal received from an external audible source , comprising:receiving a composite audible signal from each of a plurality of microphones disposed within a non-linear array of microphones, wherein the plurality of microphones comprise a first microphone, a second microphone and a third microphone; and combining at least a portion of the composite audible signal received by the first microphone with at least a portion of the composite audible signal received by the second microphone to form a first combined audible signal;', 'delaying a portion of the composite audible signal received by the third microphone a predetermined delay time;', 'inverting the delayed composite audible signal to form a first inverted audible signal; and', 'combining the first inverted audible signal and the first combined audible signal to generate the formed audible signal., 'forming an audible signal that has an improved signal-to-noise ratio as compared to ...

TWO-WAY LOUDSPEAKER WITH FLOATING WAVEGUIDE

One or more embodiments of the present disclosure relate to a two-way loudspeaker design that forces a condensed geometry between low frequency (LF) and high frequency (HF) drivers and then “floats” a midrange waveguide in front of the LF driver. This is a hybrid design meant to benefit from the close proximity of acoustic centers without introducing a central axis obstruction for the LF driver. In addition, the LF and HF waveguides and associated acoustic elements are used to redirect very low frequency energy not supported adequately by the LF waveguide to exit freely using other paths. 1. A loudspeaker comprising:a speaker enclosure;a low-frequency (LF) driver disposed in the speaker enclosure and having a radiating surface adapted to emit LF acoustic energy and a radiating surface opening defined by an outer circumference of the radiating surface; andan LF waveguide defining a first radiation path for LF acoustic energy, the LF waveguide having a proximal opening positioned adjacent to the LF driver and extending away from the LF driver to a distal opening to define the first radiation path therethough, the proximal opening having a proximal opening area that is smaller than a radiating surface opening area to define a second radiation path for the LF acoustic energy around an outer surface of the LF waveguide.2. The loudspeaker of claim 1 , wherein the distal opening of the LF waveguide has a distal opening area that is larger than the radiating surface opening area.3. The loudspeaker of claim 1 , wherein the LF waveguide is detached from the LF driver to define an air gap between the radiating surface of the LF driver and the proximal opening of the LF waveguide.4. The loudspeaker of claim 1 , wherein an inner surface and an outer surface of the LF waveguide have generally equal acoustic pressure from the LF driver.5. The loudspeaker of claim 1 , wherein the second radiation path exits the speaker enclosure along a front surface.6. The loudspeaker of claim 1 , ...

MICROPHONE MIXING FOR WIND NOISE REDUCTION

Wind noise reduction in microphone signals. A first microphone signal is obtained from a first omnidirectional microphone and, contemporaneously, a second microphone signal is obtained from a second omnidirectional microphone. The first and second microphone signals are mixed to produce an output signal. Mixing involves weighting the first and second microphone signals by respective first and second signal weights to produce respective first and second weighted microphone signals, and summing the first and second weighted microphone signals together to produce the output signal. The first and second signal weights are calculated to minimise the power of the output signal. 1. A method of wind noise reduction , the method comprisingobtaining a first microphone signal from a first omnidirectional microphone;contemporaneously obtaining a second microphone signal from a second omnidirectional microphone; and weighting the first and second microphone signals by respective first and second signal weights to produce respective first and second weighted microphone signals; and', 'summing the first and second weighted microphone signals together to produce the output signal, wherein the first and second signal weights are calculated to minimise the power of the output signal., 'mixing the first and second microphone signals to produce an output signal, by3. (canceled)4. (canceled)5. The method of wherein weights are calculated continuously for each first signal sample and second signal sample claim 1 , by calculating x claim 1 , yand xy for each sample and adding them to a respective appropriate running sum.6. The method of wherein a leaky integrator is used to perform the running sum in order to prevent overflows.7. The method of wherein the first and second signals are frequency domain samples.8. The method of wherein a weighting factor a claim 7 , is calculated for each subband i claim 7 , and the aare applied on a subband—by—subband basis to give different mixing ratios ...

Method to determine loudspeaker change of placement

A system and method is described for determining whether a loudspeaker device has relocated, tilted, rotated, or changed environment such that one or more parameters for driving the loudspeaker may be modified and/or a complete reconfiguration of the loudspeaker system may be performed. In one embodiment, the system may include a set of sensors. The sensors provide readings that are analyzed to determine 1) whether the loudspeaker has moved since a previous analysis and/or 2) a distance of movement and/or a degree change in orientation of the loudspeaker since the previous analysis. Upon determining the level of movement is below a threshold value, the system adjusts previous parameters used to drive one or more of the loudspeakers. By adjusting previous parameters instead of performing a complete recalibration, the system provides a more efficient technique for ensuring that the loudspeakers continue to produce accurate sound for the listener.

Techniques to Mitigate the Effect of Blocked Sound at Microphone Arrays in a Telepresence Device

A telepresence video conference endpoint device includes spaced-apart microphone arrays each configured to transduce sound into corresponding sound signals. A processor receives the sound signals from the arrays and determines a direction-of-arrival (DOA) of sound at each array based on the set of sound signals from that array, determines if each array is blocked or unblocked based on the DOA determined for that array, selects an array among the arrays based on whether each array is determined to be blocked or unblocked, and perform subsequent sound processing based on one or more of the sound signals from the selected array. 1. An apparatus comprising:spaced-apart microphone arrays each configured to transduce sound into corresponding sound signals; and determine a direction-of-arrival (DOA) of sound at each array based on the set of sound signals from that array;', 'determine if each array is blocked or unblocked based on the DOA determined for that array;', 'select an array among the arrays based on whether each array is determined to be blocked or unblocked; and', 'perform subsequent sound processing based on one or more of the sound signals from the selected array., 'a processor configured to receive the sound signals from the arrays and configured to2. The apparatus of claim 1 , wherein the processor is further configured to assert that each array is either unblocked or blocked claim 1 , respectively claim 1 , if the DOA determined for that array is within or not within a predetermined acceptable range of DOAs for that array.3. The apparatus of claim 2 , wherein:if it is determined that one of the arrays is unblocked and all others are blocked, the processor is configured to select the unblocked array; andif it is determined that all of the arrays are blocked, the processor is configured to select the array that was previously selected based on whether each array was previously determined to be blocked or unblocked.4. The apparatus of claim 3 , wherein: ...

Methods, Apparatus and Systems for Audio Sound Field Capture

A microphone array for capturing sound field audio content may include a first set of directional microphones disposed on a first framework at a first radius from a center and arranged in at least a first portion of a first spherical surface. The microphone array may include a second set of directional microphones disposed on a second framework at a second radius from the center and arranged in at least a second portion of a second spherical surface. The second radius may be larger than the first radius. The directional microphones may capture information that allows for the extraction of Higher-Order Ambisonics (HOA) signals.

Position directed acoustic array and beamforming methods

Methods and systems are provided for receiving desired sounds. The system includes a position sensor configured to determine an occupant position of an occupant engaging in speech within a defined space and transmit the speaking occupant position. A plurality of microphones are configured to receive sound from within the defined space and transmit audio signals corresponding to the received sound. A processor, in communication with the position sensor and the microphones, is configured to receive the speaking occupant position and the audio signals, apply a beamformer to the audio signals to direct a microphone beam toward the occupant position, and generate a beamformer output signal.

DIRECTIONAL AUDIO PICKUP IN COLLABORATION ENDPOINTS

A microphone array includes one or more front-facing microphones disposed on a front surface of the collaboration endpoint and a plurality of secondary microphones disposed on a second surface of the collaboration endpoint. The sound signals received at each of the one or more front-facing microphones and the plurality of secondary microphones are converted into microphone signals. When the sound signals have a frequency below a threshold frequency, an output signal is generated from microphone signals generated by the one or more front-facing microphones and the plurality of secondary microphones. When the sound signals have a frequency at or above a threshold frequency, an output signal is generated from microphone signals generated by only the one or more front-facing microphones. 1. An apparatus comprising:a front surface and a second surface;a microphone array including one or more front-facing microphones positioned at the front surface of the apparatus and one or more secondary microphones positioned at the second surface,wherein the one or more front-facing microphones are configured to convert received sound signals into front-facing microphone signals and the one or more secondary microphones are configured to covert received sound signals into secondary microphone signals; and 'generate, using at least one beamforming technique, a beamformer signal from the front-facing microphone signals and the secondary microphone signals.', 'one or more processors configured to2. The apparatus of claim 1 , wherein the second surface of the apparatus is substantially orthogonal to the front surface of the apparatus.3. The apparatus of claim 1 , wherein the one or more secondary microphones disposed on the second surface of the apparatus comprise a plurality of secondary microphones.4. The apparatus of claim 3 , wherein the plurality of secondary microphones comprise five microphones.5. The apparatus of claim 4 , wherein the one or more front-facing microphones comprise ...

Orientation-Based Playback Device Microphone Selection

Aspects of a multi-orientation playback device including at least one microphone array are discussed. A method may include determining an orientation of the playback device which includes at least one microphone array and determining at least one microphone training response for the playback device from a plurality of microphone training responses based on the orientation of the playback device. The at least one microphone array can detect a sound input, and the location information of a source of the sound input can be determined based on the at least one microphone training response and the detected sound input. Based on the location information of the source, the directional focus of the at least one microphone array can be adjusted, and the sound input can be captured based on the adjusted directional focus. 1. A playback device comprising:a set of microphones including a first subset of microphones and a second subset set of microphones;an orientation sensor;a network interface;a processor;a non-transitory computer-readable medium; and determining that the playback device is in a first orientation based on a first output of the orientation sensor;', 'based on the determination that the playback device is in the first orientation, selecting the first subset of microphones instead of the second subset of microphones for sound input detection;', 'detecting, via the selected first subset of microphones, a first sound input;', 'processing the first sound input detected via the selected first subset of microphones;', 'executing a first set of one or more commands contained in the processed first sound input;', 'determining that the playback device is in a second orientation based on a second output of the orientation sensor;', 'based on the determination that the playback device is in the second orientation, selecting the second subset of microphones instead of the first subset of microphones for sound input detection;', 'detecting, via the selected second subset of ...

ARRAY MICROPHONE SYSTEM AND METHOD OF ASSEMBLING THE SAME

Embodiments include a microphone assembly comprising an array microphone and a housing configured to support the array microphone and sized and shaped to be mountable in a drop ceiling in place of at least one of a plurality of ceiling tiles included in the drop ceiling. A front face of the housing includes a sound-permeable screen having a size and shape that is substantially similar to the at least one of the plurality of ceiling tiles. Embodiments also include an array microphone system comprising a plurality of microphones arranged, on a substrate, in a number of concentric, nested rings of varying sizes around a central point of the substrate. Each ring comprises a subset of the plurality of microphones positioned at predetermined intervals along a circumference of the ring. 140-. (canceled)41. A microphone system comprising:a housing;an array microphone comprising a plurality of microphones, the array microphone disposed within the housing and configured to simultaneously form a plurality of lobes at various angles;an audio processor disposed within the housing and electrically coupled to the array microphone, the audio processor configured to process audio signals captured by the plurality of microphones and generate at least one audio output; andan external port disposed within and accessible external to the housing and electrically connected to the audio processor, the external port being configured to: electrically couple a cable received therein to the audio processor and, via the cable, receive control signals from an external control system, transmit the audio output to an external audio component, and receive power from an external power supply.42. The microphone system of claim 41 , wherein the audio processor is configured to perform digital signal processing including at least one of gain control and audio mixing.43. The microphone system of claim 41 , wherein the audio processor is further configured to enable steering of a selected one of the ...

Microphone arrays and communication systems for directional reception

Disclosed herein are microphone arrays for directional reception, along with related system, devices, and techniques. For example, a four-microphone array for directional signal reception may include first, second, and third microphones arranged such that projections of the first, second, and third microphones in a plane provide corners of a triangle in the plane. In some embodiments, a fourth microphone may be arranged such that a projection of the fourth microphone in the plane is disposed in an interior of the triangle. In other embodiments, the fourth microphone may be arranged such that the projection of the fourth microphone in the plane is disposed outside the interior of the triangle, and a distance between the first microphone and the second microphone is different from a distance between the first microphone and the third microphone.

Wearable directional microphone array system and audio processing method

A wearable microphone array apparatus and system used as a directional audio system and as an assisted listening device. The present invention advances hearing aids and assisted listening devices to allow construction of a highly directional audio array that is wearable, natural sounding, and convenient to direct, as well as to provide directional cues to users who have partial or total loss of hearing in one or both ears. The advantages of the invention include simultaneously providing high gain, high directivity, high side lobe attenuation, and consistent beam width; providing significant beam forming at lower frequencies where substantial noises are present, particularly in noisy, reverberant environments; and allowing construction of a cost effective body-worn or body-carried directional audio device.

Array microphone system and method of assembling the same

Embodiments include a microphone assembly comprising an array microphone and a housing configured to support the array microphone and sized and shaped to be mountable in a drop ceiling in place of at least one of a plurality of ceiling tiles included in the drop ceiling. A front face of the housing includes a sound-permeable screen having a size and shape that is substantially similar to the at least one of the plurality of ceiling tiles. Embodiments also include an array microphone system comprising a plurality of microphones arranged, on a substrate, in a number of concentric, nested rings of varying sizes around a central point of the substrate. Each ring comprises a subset of the plurality of microphones positioned at predetermined intervals along a circumference of the ring.

AUDIO HUB AND A SYSTEM HAVING ONE OR MORE AUDIO HUBS

There may be provided a system that may include a processor and an audio hub; wherein the audio hub may include first communication interfaces, a second communication interface, a processor, and a memory; wherein the first communication interfaces may be configured to exchange audio signals with a group of audio components of different types; wherein an aggregate number of first communication interface bits exceeds a number of second communication interface bits; wherein the audio signals may include input audio signals received from the group and output audio signals transmitted to the group; wherein the processor may be configured to generate an input multiplex of input audio signals; and wherein the second communication interface may be configured to transmit the input multiplex to the processor and to receive an output multiplex from the processor. 1. A system comprising a processor and an audio hub;wherein the audio hub comprises first communication interfaces, a second communication interface, a processor, and a memory;wherein the first communication interfaces are configured to exchange audio signals with a group of audio components of different types; wherein an aggregate number of first communication interface bits exceeds a number of second communication interface bits;wherein the audio signals comprise input audio signals received from the group and output audio signals transmitted to the group;wherein the processor is configured to generate an input multiplex of input audio signals; andwherein the second communication interface is configured to transmit the input multiplex to the processor and to receive an output multiplex from the processor.2. The system according to wherein the audio hub is configured to generate the input multiplex based on a mapping stored in the audio hub.3. The system according to wherein the audio hub is configured to generate the input multiplex based on types of audio signals requested by the processor and based on a usage of ...

MULTI-MICROPHONE NEURAL NETWORK FOR SOUND RECOGNITION

A neural network is provided for recognition and enhancement of multi-channel sound signals received by multiple microphones, which need not be aligned in a linear array in a given environment. Directions and distances of sound sources may also be detected by the neural network without the need for a beamformer connected to the microphones. The neural network may be trained by knowledge gained from free-field array impulse responses obtained in an anechoic chamber, array impulse responses that model simulated environments of different reverberation times, and array impulse responses obtained in actual environments. 1. A method comprising:generating a plurality of free-field array impulse responses representing impulse responses of a plurality of microphones in free-field conditions over a plurality of directions based on sound events received by the microphones;generating a first plurality of array impulse responses that model a plurality of simulated environments of different reverberation times, wherein the simulated environments differ from the free-field conditions in which the free-field array impulse responses are generated;generating a second plurality of array impulse responses in one or more actual environments, wherein the one or more actual environments differ from the free-field conditions in which the free-field array impulse responses are generated; and applying positive training labels to input sound signals corresponding to sound events from one or more prescribed directions; and', 'applying negative training labels are applied to input sound signals corresponding to sound events from directions other than the one or more prescribed directions., 'training a neural network for sound recognition based on one or more of the free-field array impulse responses, the first plurality of array impulse responses, and the second plurality of array impulse responses, wherein training the neural network for sound recognition includes2. The method of claim 1 , ...

ACOUSTIC DEFLECTOR FOR OMNI-DIRECTIONAL SPEAKER SYSTEM

An omni-directional acoustic deflector includes an acoustically reflective body that has a truncated conical shape which includes a substantially conical outer surface that is configured to be disposed adjacent an acoustically radiating surface (e.g., a diaphragm) of an acoustic driver thereby to define an acoustic radiation path therebetween. The acoustically reflective body is profiled such that a cross-sectional area of the acoustic radiation path increases monotonically with respect to radial distance from a motion axis of the acoustic driver. 1. An omni-directional acoustic deflector , comprising:an acoustically reflective body having a truncated conical shape including a substantially conical outer surface configured to be disposed adjacent an acoustically radiating surface of an acoustic driver thereby to define an acoustic radiation path therebetween,wherein the acoustically reflective body is profiled such that a cross-sectional area of the acoustic radiation path increases monotonically with respect to radial distance from a motion axis of the acoustic driver.2. The omni-directional acoustic deflector of claim 1 , wherein the substantially conical outer surface has a steeper slope than the acoustically radiating surface.3. The omni-directional acoustic deflector of claim 1 , wherein the substantially conical outer surface has a linear slant profile wherein the cross-sectional area of the acoustic radiation path increases in linear fashion with respect to distance from a motion axis of the acoustic driver.4. The omni-directional acoustic deflector of claim 1 , wherein the substantially conical outer surface has a non-linear slant profile wherein the cross-sectional area of the acoustic radiation path increases in a nonlinear fashion with respect to distance from a motion axis of the acoustic driver.5. The omni-directional acoustic deflector of claim 4 , wherein the substantially conical outer surface has a substantially parabolic profile claim 4 , wherein ...

DIFFERENTIAL MICROPHONE WITH DUAL POLARITY BIAS

Methods and system are described for cancelling interference in a microphone system. A positive bias voltage is applied to a first microphone diaphragm and a negative bias voltage is applied to a second microphone diaphragm. The diaphragms are configured to exhibit substantially the same mechanical deflection in response to acoustic pressures received by the microphone system. A differential output signal is produced by combining a positively-biased output signal from the first microphone diaphragm and a negatively-biased output signal from the second microphone diaphragm. This combining cancels common-mode interferences that are exhibited in both the positively-biased output signal and the negatively-biased output signal. 1. A microphone system including:a first microphone diaphragm positioned to receive acoustic pressure from a first direction, wherein a positive bias voltage is applied to the first diaphragm;a second microphone diaphragm positioned to receive acoustic pressure from the first direction at substantially the same amplitude as the acoustic pressure received by the first microphone diaphragm, wherein a negative bias voltage is applied to the first diaphragm; and receive a positively-biased output from the first microphone diaphragm and a negatively-biased output from the second microphone diaphragm, and', 'produce a differential signal based on the positively-biased output and the negatively-biased output., 'a differential amplifier configured to'}2. A method of cancelling interference in a microphone system , the method comprising:applying a positive bias voltage to a first microphone diaphragm;applying a negative bias voltage to a second microphone diaphragm, the second microphone diaphragm being configured to exhibit substantially the same mechanical deflection as the first microphone diaphragm in response to acoustic pressures received by the microphone system; andproducing a differential output signal by combining a positively-biased output ...

Aerial acoustic sensing, acoustic sensing payload and aerial vehicle including the same

An aerial acoustic acquisition system including: an unmanned aerial vehicle (UAV); an acoustic sensing payload attached to the UAV including: at least one SOI microphone configured to detect a first audio signal including a signal of interest; and at least one noise detection microphone configured to detect a second audio signal including sound generated by the UAV, and a processing suite including a processor configured to receive first audio data corresponding to the first audio signal and second audio data corresponding to the second audio signal from the acoustic sensing suite, and process the first audio data using the second audio data to extract the signal of interest from the first audio data.

Flexible Spatial Audio Capture Apparatus

An apparatus comprising: at least one determiner configured to determine at least one characteristic associated with a flexible part of a further apparatus; a signal processor configured to process at least one signal dependent on the at least one characteristic associated with the flexible part of the further apparatus, wherein the signal is at least one of an audio and a video signal; and an user interface generator configured to generate at least one user interface indication dependent on the characteristic associated with a flexible part of the further apparatus. 128-. (canceled)29. An apparatus comprising:at least one processor, and receiving at least one audio signal generated from one or more microphones associated with a further apparatus;', 'determining at least one bend characteristic corresponding to a flexible part of the further apparatus; and', 'processing the at least one audio signal based on the at least one bend characteristic to define a width of an audio capture field of the one or more microphones., 'at least one non-transitory memory including computer code for one or more programs, the at least one memory and the computer code configured to, with the at least one processor, cause the apparatus to perform operations comprising30. The apparatus as in claim 29 , wherein the at least one bend characteristic is indicative of at least one of:a bend direction of the flexible part of the further apparatus; anda value indicative of an amount the flexible part is bent relative to a flat position of the flexible part of the further apparatus.31. The apparatus as in claim 29 , wherein the operations further comprise:adjusting the width of the audio capture field in response to detecting a change in the at least one bend characteristic.32. The apparatus as in claim 31 , wherein adjusting the width of the audio capture field of the one or more microphones comprises either broadening the audio capture field based on the at least one bend characteristic claim ...

ARRAY MICROPHONE SYSTEM AND METHOD OF ASSEMBLING THE SAME

Embodiments include a microphone assembly comprising an array microphone and a housing configured to support the array microphone and sized and shaped to be mountable in a drop ceiling in place of at least one of a plurality of ceiling tiles included in the drop ceiling. A front face of the housing includes a sound-permeable screen having a size and shape that is substantially similar to the at least one of the plurality of ceiling tiles. Embodiments also include an array microphone system comprising a plurality of microphones arranged, on a substrate, in a number of concentric, nested rings of varying sizes around a central point of the substrate. Each ring comprises a subset of the plurality of microphones positioned at predetermined intervals along a circumference of the ring. 140-. (canceled)41. An array microphone system comprising:a plurality of microphones arranged in a number of concentric, nested configurations, each configuration comprising a subset of the plurality of microphones positioned at predetermined intervals from each other to form the configuration, wherein the configurations are rotationally offset from each other relative to a central axis of the array microphone system.42. The array microphone system of claim 41 , wherein each configuration is rotationally offset from the central axis by a different number of degrees.43. The array microphone system of claim 41 , wherein each configuration is positioned at a different radial distance from the central axis to form the nested arrangement.44. The array microphone system of claim 41 , further comprising at least one microphone arranged at a central point of the concentric claim 41 , nested configurations.45. The array microphone system of claim 41 , wherein each configuration forms a circular shape with a diameter that is selected based on a lowest operating frequency assigned to the subset of microphones included in that configuration.46. The array microphone system of claim 41 , wherein each ...

MICROPHONE-BASED ORIENTATION SENSORS AND RELATED TECHNIQUES

An orientation detector can have a first microphone, a second microphone, and a reference microphone spaced from the first microphone and the second microphone. An orientation processor can be configured to determine an orientation of the first microphone, the second microphone, or both, relative to a user's mouth based on a comparison of a relative strength of a first signal associated with the first microphone to a relative strength of a second signal associated with the second microphone. A channel selector in a speech enhancer can select one signal from among several signals based at least in part on the orientation determined by the orientation processor. A mobile communication handset can include a microphone-based orientation detector of the type disclosed herein. 1. An orientation detector comprising:a first microphone having first position, a second microphone having a second position, and a reference microphone spaced from the first microphone and the second microphone;an orientation processor configured to determine an orientation of the first microphone, the second microphone, or both, relative to a user's mouth based on a comparison of a relative separation of a first signal associated with the first microphone to a relative separation of a second signal associated with the second microphone.2. The orientation detector according to claim 1 , wherein the first signal comprises a signal emitted by the first microphone transducer.3. The orientation detector according to claim 2 , wherein the first signal further comprises a combination of the signal emitted by the first microphone with a signal emitted by the second microphone claim 2 , wherein at least a portion of the signal emitted by the first microphone transducer is more heavily weighted in the combination relative to at least a portion of the signal emitted by the second microphone transducer.4. The orientation detector according to claim 2 , wherein the second signal comprises a signal emitted by ...

Acoustic device

An acoustic device that is adapted to be worn on the body of a user, with a first acoustic transducer and a second acoustic transducer, where the first transducer is closer to the expected location of a first ear of the user than is the second transducer, a third acoustic transducer and a fourth acoustic transducer, where the third transducer is closer to the expected location of a second ear of the user than is the fourth transducer, and a controller that is adapted to independently control the phase and frequency response of the first, second, third and fourth transducers.

Loudspeaker System

Provided is a loudspeaker system capable of easily determining control parameters for controlling the directional characteristics of a speaker set including two or more speaker units. The loudspeaker system is configured to include: a speaker set including two or more speaker units a camera adapted to photograph an acoustic space of the speaker set a monitor adapted to display a camera image photographed by the camera and a directional control parameter generating part adapted to generate directional control parameters providing the directional characteristics of the speaker set The directional control parameter generating parts includes: a target position designation part adapted to, on the basis of user operations, designate target positions on the camera image a directional control angle calculation part adapted to, on the basis of the target positions obtain directional control angles θ with respect to a front direction of the speaker set and a directional control parameter determination part adapted to, on the basis of the directional control angles θ, determine the directional control parameters. 1. A loudspeaker system comprising:a speaker set including two or more speaker units;a camera that photographs an acoustic space of said speaker set, and has a predetermined relative positional relationship with said speaker set;camera image display means adapted to display a camera image photographed by said camera;target position designation means adapted to, on a basis of a user operation, designate one or more than one target position on said camera image;directional control angle calculation means adapted to, on a basis of said target position, obtain one or more than one directional control angle with respect to a front direction of said speaker set; anddirectional control means adapted to, on a basis of said directional control angle, perform directional control of said speaker set.2. The loudspeaker system according to claim 1 , whereinsaid directional control ...

SPEAKER SYSTEM WHICH COMPRISES SPEAKER DRIVER GROUPS

A speaker system (″) comprises a first speaker driver group (G), comprising at least one first speaker driver (S) and a first amplifier (A), and a second speaker driver group (G), comprising at least one second speaker driver (S) and a second amplifier (A). The speaker system further comprises a digital signal processor (), adapted to provide a first signal to the first speaker driver group (G) and a second signal to the second signal group (G), wherein the first and second signals differ with respect to frequency range and at least one of the speaker driver groups comprises at least 4, 6, 8 or 10 speaker drivers. 127.-. (canceled)28. A speaker system , comprising:a cabinet; anda one-dimensional array of speaker drivers, arranged in the cabinet,wherein the one-dimensional array of speaker drivers comprises at least one central speaker driver configured to play higher frequencies, and a plurality of outer speaker drivers configured to play lower frequencies.29. A speaker system according to claim 28 , comprising more outer speaker drivers than central speaker drivers.30. A speaker system according to claim 28 , wherein a first group of the plurality of outer speaker drivers are located on a first side of the at least one central speaker driver claim 28 , and a second group of the outer speaker drivers are located on a second side of the central speaker driver claim 28 , wherein the second side is opposite the first side.31. A speaker system according to claim 28 , wherein the plurality of outer speaker drivers configured to play lower frequencies comprises:a first plurality of outer speaker drivers located at first distances from the at least one central speaker driver, anda second plurality of outer speaker drivers located at second distances from the at least one central speaker driver,wherein the second distances are greater than the first distances, and the second plurality of outer speaker drivers are configured to play lower frequencies than the first plurality ...

MICROPHONE MIXING FOR WIND NOISE REDUCTION

Wind noise reduction in microphone signals. A first microphone signal is obtained from a first omnidirectional microphone and, contemporaneously, a second microphone signal is obtained from a second omnidirectional microphone. The first and second microphone signals are mixed to produce an output signal. Mixing involves weighting the first and second microphone signals by respective first and second signal weights to produce respective first and second weighted microphone signals, and summing the first and second weighted microphone signals together to produce the output signal. The first and second signal weights are calculated to minimize the power of the output signal. 1. A method of wind noise reduction , the method comprisingobtaining a first microphone signal from a first omnidirectional microphone;contemporaneously obtaining a second microphone signal from a second omnidirectional microphone; and weighting the first and second microphone signals by respective first and second signal weights to produce respective first and second weighted microphone signals; and', 'summing the first and second weighted microphone signals together to produce the output signal,, 'mixing the first and second microphone signals to produce an output signal, bywherein the first and second signal weights are calculated to minimize the power of the output signal.3. (canceled)4. (canceled)5. The method of wherein weights are calculated continuously for each first signal sample and second signal sample claim 1 , by calculating x claim 1 , yand xy for each sample and adding them to a respective appropriate running sum.6. The method of wherein a leaky integrator is used to perform the running sum in order to prevent overflows.7. The method of wherein the first and second signals are frequency domain samples.8. The method of wherein a weighting factor ais calculated for each subband i claim 7 , and the aare applied on a subband-by-subband basis to give different mixing ratios at different ...

MICROPHONE ASSEMBLY HAVING A RECONFIGURABLE GEOMETRY

The disclosure relates to a microphone assembly () for acquiring a plurality of audio signals, wherein the microphone assembly () has a reconfigurable geometry so that the microphone assembly () may be configured to be embedded in or attached to a body. The microphone assembly () comprises: a plurality of digital microphones (-) configured to convert the sound signal impinging on each digital microphone into a corresponding digital electrical signal, a digital signal processing unit () comprising a serial digital communication interface () and a processor (), and a connecting and mounting structure configured to provide a flexible electrical connection and a flexible mechanical arrangement for the plurality of digital microphones (-). 1. A microphone assembly for acquiring a plurality of audio signals , the microphone assembly having a reconfigurable geometry so that the microphone assembly may be configured to be embedded in or attached to a body , the microphone assembly comprising:a plurality of digital microphones configured to convert a sound signal impinging on each of the digital microphones into a corresponding digital electrical signal, wherein the digital electrical signal is a Pulse-Code-Modulation (PCM) or a Pulse-Density-Modulation (PDM) digital serial electrical signal;a digital signal processor comprising a serial digital communication interface and a processor, wherein the serial digital communication interface is configured to receive the digital electrical signals provided by the plurality of digital microphones, and wherein the processor is configured to store, transmit, and/or process the digital electrical signals provided by the plurality of digital microphones; anda connecting and mounting structure configured to provide a flexible electrical connection and a flexible mechanical arrangement for the plurality of digital microphones, wherein the connecting and mounting structure comprises a plurality of wired connections configured to connect ...

Endfire linear array microphone

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

Compact surround-sound effects system

A loudspeaker suitable for the generation of surround-sound effects comprising a plurality of transducers that are together arranged to generate one or more beams of sound, at least one of which beams is steerable, wherein the transducers are arranged to reproduce sound only at frequencies higher than a selected lower frequency limit of 250 Hz or more. The loudspeaker may be foldable or dismantable so as to allow greater portability and may have means for interacting with an external electronics device, for example to allow low frequency sound signals to be reproduced by woofers in the external device.

ARRAY MICROPHONE SYSTEM AND METHOD OF ASSEMBLING THE SAME

Embodiments include a microphone assembly comprising an array microphone and a housing configured to support the array microphone and sized and shaped to be mountable in a drop ceiling in place of at least one of a plurality of ceiling tiles included in the drop ceiling. A front face of the housing includes a sound-permeable screen having a size and shape that is substantially similar to the at least one of the plurality of ceiling tiles. Embodiments also include an array microphone system comprising a plurality of microphones arranged, on a substrate, in a number of concentric, nested rings of varying sizes around a central point of the substrate. Each ring comprises a subset of the plurality of microphones positioned at predetermined intervals along a circumference of the ring. 140-. (canceled)41. A microphone system comprising:an array microphone comprising a plurality of microphones, the array microphone configured to simultaneously form a plurality of lobes at various angles to capture a plurality of audio sources;an audio processor electrically coupled to the array microphone, the audio processor configured to process audio signals captured by the plurality of microphones and generate at least one audio output based on the processed audio signals;a housing configured to encase the array microphone and the audio processor; anda port coupled to the housing and electrically connected to the audio processor, the port configured to: receive control signals from an external control system, transmit the at least one audio output to an external audio component, and receive power from an external power supply.42. The microphone system of claim 41 , wherein the housing is sized and shaped to be mountable in a drop ceiling in place of at least one of a plurality of ceiling tiles included in the drop ceiling.43. The microphone system of claim 41 , wherein the housing comprises:a first face that includes a sound-permeable screen, the screen having a size and shape that is ...

Multi-channel multi-domain source identification and tracking

An audio source location, tracking and isolation system, particularly suited for use with person-mounted microphone arrays. The system increases capabilities by reducing resources required for certain functions so those resources can be utilized for result enhancing processes. A wide area scan may be utilized to identify the general vicinity of an audio source and a narrow scan to locate pinpoint positions may be initiated in the general vicinity identified by the wide area scan. Subsequent locations may be anticipated by compensating for motion of the sensor array and anticipated changes in source location by trajectory. Identification may use two or more sets of characterizations and rules. The characterizations may use computationally less intense analyzes to characterize audio and only perform computationally higher intensity analysis if needed. Rule sets may be used to eliminate the need to track audio sources that emit audio to be eliminated from an audio output.

Direction detection device for acquiring and processing audible input

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.

Device for acquiring and processing audible input

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.

具有改进方向性的二维麦克风阵列

实施例包含一种阵列麦克风，其包括多个麦克风组，所述多个麦克风组以相对于第一轴的线性模式布置并且经配置以覆盖多个频带。每一麦克风组包括沿所述第一轴布置的第一麦克风及沿与所述第一麦克风正交的第二轴布置的第二麦克风，其中邻近麦克风之间沿所述第一轴的距离从由第一值的整数倍组成的第一群组选择，并且在每一元件内，所述第一麦克风与所述第二麦克风之间沿所述第二轴的距离从由第二值的整数倍组成的第二群组选择。

Systems, methods, and apparatus for enhanced creation of an acoustic image space

음향 심리적 저음 향상 신호를 이용하여 라우드스피커들의 어레이를 구동하기 위한 방법들, 시스템들 및 장치들이 개시된다. Methods, systems and apparatus for driving an array of loudspeakers using acoustic psycho-acoustic enhancement signals are disclosed.

System for a sonar device and underwater vehicle and method using same

Sound emission and collection device

Circularly symmetric, zero redundancy, planar array having broad frequency range applications

一类平面阵列,具有源定位,源成像或以投射波束作目标照射等宽频率范围应用。该非冗余阵列是圆对称的并由多个传感或发送单元组成,这些单元的排列在宽频率范围内可大体消除栅瓣。从各单元接收或发送给各单元的信号被适当相控以控制阵列的波束。

One-dimensional array microphone with improved directivity

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

Endfire linear array microphone

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

Array microphone system and method of assembling the same

Embodiments include a microphone assembly comprising an array microphone and a housing configured to support the array microphone and sized and shaped to be mountable in a drop ceiling in place of at least one of a plurality of ceiling tiles included in the drop ceiling. A front face of the housing includes a sound-permeable screen having a size and shape that is substantially similar to the at least one of the plurality of ceiling tiles. Embodiments also include an array microphone system comprising a plurality of microphones arranged, on a substrate, in a number of concentric, nested rings of varying sizes around a central point of the substrate. Each ring comprises a subset of the plurality of microphones positioned at predetermined intervals along a circumference of the ring.

Adjustable lobe shape for array microphones

Array microphone systems and methods having adjustable lobe shapes are provided. The lobe shapes of pickup patterns in an array microphone may be adjusted by weighting the audio signals of subsets of the microphone elements that make up the array. The lobe shapes may be adjusted in a direction independent of a steering vector of the lobe. Users may have greater control of lobes which can result in more efficient and optimal coverage of audio sources in environments.

Steerable speaker array, system, and method for the same

A steerable speaker array is provided, comprising a plurality of drivers arranged in a concentric, nested configuration formed by arranging the drivers in a plurality of concentric groups and placing the groups at different radial distances from a central point of the configuration. Each group is formed by a subset of the plurality of drivers being positioned at predetermined intervals from each other along a perimeter of the group. Also, the concentric groups are harmonically nested and rotationally offset from each other. An audio system is also provided comprising at least one steerable speaker array and a beamforming system configured to receive one or more input audio signals from an audio source, generate a separate audio output signal for each driver of the speaker array based on at least one of the input signals, and provide the audio output signals to the corresponding drivers to produce a beamformed audio output.

Acoustic sensor assembly and method for sensing sound using the same

음향 센서 어셈블리는 제1 지향 패턴을 갖는 무지향성(omni-directional) 음향 센서, 무지향성 음향 센서를 포위하도록 배치되고, 각각 서로 다른 공진 주파수를 가지는 복수의 공진기들을 포함하고, 각각 제2 지향 패턴을 갖는 복수의 지향성(directional) 음향 센서들 및 무지향성 음향 센서 및 복수의 지향성 음향 센서들로부터 출력신호들을 획득하고, 상기 출력신호들 중 적어도 하나를 선택하거나 선택적으로 조합함으로써 지향성을 갖는 음향 신호를 산출하고, 음향 신호를 이용하여 음향 센서 어셈블리 주변의 음향을 획득하는 프로세서를 포함할 수 있다.

Conferencing device self test

A plurality of acoustic sensors in a non-anechoic environment are calibrated with the aim of removing manufacturing tolerances and degradation over time but preserving position-dependent differences between the sensors, The sensors are excited by an acoustic stimulus which has either time-dependent characteristics or finite duration. The calibration is to be based on diffuse-field excitation only, in which indirect propagation (including single or multiple reflections) dominate over any direct-path excitation. For this purpose, the calibration process considers only a non-initial portion of sensor outputs and/or of an impulse response derived there-from. Based on these data, a frequency-dependent magnitude response function is estimated and compared with a target response function, from which a calibration function is derived.

Surround speaker system

스피커 시스템이 개시된다. 개시된 스피커 시스템은, 청취자의 전방으로 그 중앙에 배치된 센터 스피커와, 상기 센터 스피커의 일측에 배치된 프론트 R스피커와, 상기 센터 스피커의 타측에 배치된 프론트 L스피커를 구비하여 된 프론트 스피커부와, 상기 프론트 R스피커의 외측으로 배치된 리어 R스피커와, 상기 프론트 L스피커의 외측으로 배치된 리어 L스피커를 구비하여 된 리어 스피커부를 포함하여 된 스피커 시스템에 있어서, 상기 프론트 스피커부와 상기 리어 스피커부는 각각 별도로 구비되는 동시에 소정의 연결수단에 의해 연결된 것을 그 특징으로 한다. A speaker system is disclosed. The disclosed speaker system includes a front speaker unit including a center speaker disposed at the center of the front of the listener, a front R speaker disposed at one side of the center speaker, and a front L speaker disposed at the other side of the center speaker; And a rear speaker unit including a rear R speaker arranged outside the front R speaker and a rear L speaker arranged outside the front L speaker, wherein the front speaker unit and the rear speaker are provided. Each part is provided separately and is characterized in that connected by a predetermined connection means. 본 발명에 따르면, 설치가 간단하고, 종래와 같이 리어 스피커 설치에 따른 배선이 필요 없어 편리성이 우수하고, 케이블이 별도로 소요되지 않으며, 리어 스피커부가 다수(Array)로서 동작하고 지향성을 강화할 수 있는 이점이 있다. According to the present invention, the installation is simple, and there is no need for wiring according to the rear speaker installation as in the prior art, and the convenience is excellent, and the cable is not required separately, and the rear speaker unit can operate as an array and reinforce the directivity. There is an advantage. 프론트 스피커, 리어 스피커, 센터 스피커 Front speakers, rear speakers, center speakers