Entries |
Document | Title | Date |
20080199023 | Assembly, System and Method for Acoustic Transducers - The invention relates to an assembly of acoustic transducers, a system and a method for receiving and reproducing sound. The assembly comprises a first acoustic transducer having a directional pattern of the shape of a figure of eight in the direction of an X axis of a XYZ coordinate system, and a second acoustic transducer placed perpendicularly relative to a first capsule and providing a directional pattern of the shape of a figure of eight in the direction of a Y axis of a XYZ coordinate system. The assembly is characterized in that it further comprises a third acoustic transducer placed perpendicularly relative to the first and second acoustic transducers, enabling the implementation of spatial sound both in a XY plane and in a XYZ plane by using these acoustic transducers placed in accordance with an axis of the axes of the XYZ coordinate system. The invention further provides a system and a method for processing signals received with the assembly. | 08-21-2008 |
20080199024 | Sound source characteristic determining device - There is provided a sound source characteristic determining device ( | 08-21-2008 |
20080199025 | SOUND RECEIVING APPARATUS AND METHOD - A plurality of sound receiving units is installed onto an equipment body. An initial information memory stores an initial direction of the equipment body in a terminal coordinate system based on the equipment body. An orientation detection unit detects an orientation of the equipment body in a world coordinate system based on a real space. A lock information output unit outputs lock information representing to rock the orientation. An orientation information memory stores the orientation detected when the lock information is output. A direction conversion unit converts the initial direction to a target sound direction in the world coordinate system by using the orientation stored in the orientation information memory. A directivity forming unit forms a directivity of the plurality of sound receiving units toward the target sound direction. | 08-21-2008 |
20080205665 | VOICE CONFERENCE APPARATUS - A voice conference apparatus includes a sound collecting unit and a loudspeaker, while the sound collecting unit has a directional polar sensitivity characteristic which has a higher sensitivity with respect to sounds which are radiated from at least one direction, as compared with sounds radiated from other directions. The sound collecting unit of the voice conference apparatus has a plurality of omnidirectional microphones, and forms a desirable sensitivity characteristic. Since the omnidirectional microphones are employed, aging changes and fluctuations contained in the sensitivity characteristics of the respective sound collecting units can be reduced, so that the sensitivity characteristics thereof can become stable, and thus, full duplex communications with higher qualities can be carried out. | 08-28-2008 |
20080219469 | Full Range Planar Magnetic Microphone And Arrays Thereof - Contemplated planar magnetic microphones have a magnet and diaphragm arrangement such that substantially homogenous vertical and high horizontal magnetic flux density is realized in the inter-magnet space. Most preferably, the diaphragm is disposed in the inter-magnet space and includes a voice coil covering a significant fraction of the active portion of the membrane. In further especially preferred aspects, the membrane is sufficiently strong and tensioned to allow a large elastic excursion in the inter-magnet space. Consequently, contemplated planar magnetic microphones provide exceptionally large dynamic range without compression and/or distortion and can be easily configured to operate in an environment that is subject to moisture, rain, or to even operate in a submerged environment. Moreover, contemplated microphones can be used as speakers at even high SPL without reconfiguration. | 09-11-2008 |
20080219470 | Signal processing apparatus, signal processing method, and program recording medium - A signal processing apparatus includes a receiving unit configured to receive an audio signal, and a noise reducing unit configured to reduce a wind noise component of the audio signal received by the receiving unit by reducing a signal component that has a frequency less than or equal to a predetermined frequency and that is localized in a different manner from a specified manner. | 09-11-2008 |
20080240463 | Enhanced Beamforming for Arrays of Directional Microphones - A novel enhanced beamforming technique that improves beamforming operations by incorporating a model for the directional gains of the sensors, such as microphones, and provides means of estimating these gains. The technique forms estimates of the relative magnitude responses of the sensors (e.g., microphones) based on the data received at the array and includes those in the beamforming computations. | 10-02-2008 |
20080247565 | Position-Independent Microphone System - An audio system generates position-independent auditory scenes using harmonic expansions based on the audio signals generated by a microphone array. In one embodiment, a plurality of audio sensors are mounted on the surface of a sphere. The number and location of the audio sensors on the sphere are designed to enable the audio signals generated by those sensors to be decomposed into a set of eigenbeam outputs. Compensation data corresponding to at least one of the estimated distance and the estimated orientation of the sound source relative to the array are generated from eigenbeam outputs and used to generate an auditory scene. Compensation based on estimated orientation involves steering a beam formed from the eigenbeam outputs in the estimated direction of the sound source to increase direction independence, while compensation based on estimated distance involves frequency compensation of the steered beam to increase distance independence. | 10-09-2008 |
20080247566 | Sound source localization system and sound source localization method - A sound source localization system and a sound source localization method. The sound source localization system includes sound capturing devices and an arithmetic unit. The sound capturing devices sense a sound source to output time domain signals. The arithmetic unit transforms the time domain signals into frequency domain signals, performs a cross spectrum process according to the frequency domain signals to determine time differences of arrival, and locates the sound source according to the time differences of arrival and locations of the sound capturing devices. | 10-09-2008 |
20080247567 | Directional Audio Capturing - Method and system for digitally directive focusing and steering of sampled sound within a target area for producing a selective audio output accompanying video. In a preferred embodiment, the method and system is characterized by receiving position and focus data from one or more cameras shooting an event, and use this input data for generating relevant sound output together with the picture. | 10-09-2008 |
20080253583 | ALWAYS ON HEADWEAR RECORDING SYSTEM - At least one exemplary embodiment is directed to an earpiece that records audio and stores the recording for a period of time. | 10-16-2008 |
20080260178 | AUDIO SIGNAL TRANSMISSION/RECEPTION DEVICE AND MICROPHONE APPARATUS THEREOF - An audio signal transmission/reception device includes a speaker array having a plurality of linearly arranged speaker units and a microphone apparatus having a microphone array having a plurality of linearly arranged microphone units. Some of the microphone units are aligned with equal spacing corresponding to a prescribed distance therebetween in a high-density alignment section, which is set symmetrical to an alignment origin corresponding to a center point of linear alignment. The remaining microphone units are aligned in a low-density alignment section externally of the high-density alignment section in such a way that the spacing therebetween is progressively widened integer times larger than the prescribed distance. Manufacturing costs can be reduced by reducing the total number of the microphone units, and it is possible to improve sound reception directivity with respect to both high and low frequency bands. | 10-23-2008 |
20080267422 | Microphone Array and Digital Signal Processing System - A digital microphone array is configured in an open geometry such as a sphere with a large number of inexpensive microphone elements mounted in opposite-facing pairs. The microphone array with DSP is intended to be placed in a three-dimensional sound field, such as a concert hall or film location, and to completely isolate all sound sources from each other while maintaining their placement in a coherent sound field including reverberance. | 10-30-2008 |
20080267423 | Object sound extraction apparatus and object sound extraction method - An object sound extraction apparatus includes sound source separation sections for separating and generating an object sound separation signal corresponding to an object sound and reference sound separation signals corresponding to the other reference sound based on each combination of a main acoustic signal and sub acoustic signals, an object sound separation signal synthesis section for synthesizing the object sound separation signals, and a spectrum subtraction processing section for extracting an acoustic signal corresponding to the object sound from the synthesis signal by performing a spectrum subtraction processing between the synthesis signal and the reference sound separation signals. Accordingly, in acoustic environments where the object sound and the noises are mixed in the acoustic signals obtained via the microphones, and the mixed conditions can vary, a high object sound extraction performance can be ensured by a small object sound extraction apparatus. | 10-30-2008 |
20080279391 | MICROPHONE UNIT AND SOUND SOURCE DIRECTION IDENTIFICATION SYSTEM - A microphone unit is provided to minimize the attenuation levels of received sound information, which differs depending upon the distance between the positions of microphones and a sound source. A sound source direction identification system is provided to identify the sound source direction. In addition, a moving head control system is provided, where the moving head control system includes a microphone system for receiving sound from a sound source, a sound source direction identification section for identifying the direction of the sound source by obtaining received sound information, a motor control section for generating an appropriate control command to a head moving motor, and a head moving motor for receiving the control command from the motor control section and moving or rotating a robot head in a direction according to the command. | 11-13-2008 |
20080285770 | SERIALLY CONNECTED MICROPHONES - The invention provides a microphone. The microphone receives a first sound signal and at least one second electrical signal and outputs a third electrical signal. In one embodiment, the microphone comprises a transducer and a signal processor. The transducer converts the first sound signal to a first electrical signal. The signal processor has a first input terminal receiving the first electrical signal and at least one second input terminal receiving the at least one second electrical signal, and derives the third electrical signal from the first electrical signal and the second electrical signal. In one embodiment, the at least one second electrical signal is derived from a t least one second sound signal by at least one second microphone located in the vicinity of the microphone. In another embodiment, the at least one second electrical signal comprises a wind noise signal derived from wind pressure by a pressure sensor located in the vicinity of the microphone. | 11-20-2008 |
20080285771 | Teleconferencing Apparatus - A teleconferencing apparatus includes the functions of a transmitting unit and a receiving unit and the transmitting unit transmits a sound signal formed from sound pick-up signals of a microphone array made up of microphones Mi (i=1 to N) and position information. The position information is provided by forming a plurality of sound pick-up beams directed in a specific direction and selecting the sound pick-up beam with the largest volume. In the receiving unit, a parameter calculation section sets a virtual sound source based on data of a reception signal and sets a delay parameter. A virtual sound source generation signal processing section forms a sound emission beam based on the parameters and outputs the beam to a loudspeaker SPi. | 11-20-2008 |
20080285772 | ACOUSTIC LOCALIZATION OF A SPEAKER - A system locates a speaker in a room containing a loudspeaker and a microphone array. The loudspeaker transmits a sound that is partly reflected by a speaker. The microphone array detects the reflected sound and converts the sound into a microphone signal. A processor determines the speaker's direction relative to the microphone array, the speaker's distance from the microphone array, or both, based on the characteristics of the microphone signals. | 11-20-2008 |
20080310649 | Sound collector and sound recorder - A sound collector includes a first microphone unit and a second microphone unit having a single directivity and being pivotally supported in a manner that directions of directional axes of the units are changeable in an identical flat plane and a switch to be controlled in conjunction with the rotations of the first and the second microphone units. Output signals of the first and the second microphone units are outputted with channels of the signals being exchanged or non-exchanged by the switch in accordance with an angle formed by the directional axes. | 12-18-2008 |
20080317259 | METHOD AND APPARATUS FOR NOISE SUPPRESSION IN A SMALL ARRAY MICROPHONE SYSTEM - A small array microphone system includes an array microphone having a plurality of microphones and operative to provide a plurality of received signals, each microphone providing one received signal. A first voice activity detector (VAD) provides a first voice detection signal generated using the plurality of received signals to indicate the presence or absence of in-beam desired speech. A second VAD provides a second voice detection signal generated using the plurality of received signals to indicate the presence or absence of out-of-beam noise when in-beam desired speech is absent. A reference signal generator provides a reference signal based on the first voice detection signal, the plurality of received signals, and a beamformed signal, wherein the reference signal has the desired speech suppressed. A beamformer provides the beamformed signal based on the second voice detection signal, the reference signal, and the plurality of received signals, wherein the beamformed signal has noise suppressed. A multi-channel noise suppressor operative to further suppress noise in the beamformed signal and provide an output signal. A speech reliability detector provides a reliability detection signal indicating the reliability of each frequency subband. The first voice detection signal, the second voice detection signal, the reliability detection signal and the output signal are provided to the speech recognition engine. | 12-25-2008 |
20080317260 | SOUND DISCRIMINATION METHOD AND APPARATUS - A method of distinguishing sound sources includes the step of transforming data, collected by at least two transducers which each react to a characteristic of an acoustic wave, into signals for each transducer location. The transducers are separated by a distance of less than about 70 mm or greater than about 90 mm. The signals are separated into a plurality of frequency bands for each transducer location. For each band a comparison is made of the relationship of the magnitudes of the signals for the transducer locations with a threshold value. A relative gain change is caused between those frequency bands whose magnitude relationship falls on one side of the threshold value and those frequency bands whose magnitude relationship falls on the other side of the threshold value. As such, sound sources are discriminated from each other based, on their distance from the transducers. | 12-25-2008 |
20090003621 | SOUND-DIRECTION DETECTOR HAVING A MINIATURE SENSOR - A representative embodiment of the invention provides a sound-direction detector having a miniature sensor coupled to a signal-processing block. The sensor has (i) a microphone responsive to a sound wave and (ii) a differential pressure sensor (DPS) responsive to a pressure difference induced by the sound wave between two inlet ports located in proximity to the microphone. The signal-processing block applies phase-sensitive detection to the output signal generated by the DPS, while using the output signal generated by the microphone as a reference for the phase-sensitive detection, to measure the pressure difference. The signal-processing block then determines direction to the sound-wave source based on the amplitude of the sound wave at the microphone and the measured pressure difference. | 01-01-2009 |
20090003622 | Advanced Speech Encoding Dual Microphone Configuration (DMC) - A microphone array is described for use in ultra-high acoustical noise environments. The microphone array includes two directional close-talk microphones. The two microphones are separated by a short distance so that one microphone picks up more speech than the other. The microphone array can be used along with an adaptive noise removal program to remove a significant portion of noise from a speech signal of interest. | 01-01-2009 |
20090003623 | Dual Omnidirectional Microphone Array (DOMA) - A dual omnidirectional microphone array noise suppression is described. Compared to conventional arrays and algorithms, which seek to reduce noise by nulling out noise sources, the array of an embodiment is used to form two distinct virtual directional microphones which are configured to have very similar noise responses and very dissimilar speech responses. The only null formed is one used to remove the speech of the user from V | 01-01-2009 |
20090003624 | Dual Omnidirectional Microphone Array (DOMA) - A dual omnidirectional microphone array noise suppression is described. Compared to conventional arrays and algorithms, which seek to reduce noise by nulling out noise sources, the array of an embodiment is used to form two distinct virtual directional microphones which are configured to have very similar noise responses and very dissimilar speech responses. The only null formed is one used to remove the speech of the user from V | 01-01-2009 |
20090003625 | Dual Omnidirectional Microphone Array (DOMA) - A dual omnidirectional microphone array noise suppression is described. Compared to conventional arrays and algorithms, which seek to reduce noise by nulling out noise sources, the array of an embodiment is used to form two distinct virtual directional microphones which are configured to have very similar noise responses and very dissimilar speech responses. The only null formed is one used to remove the speech of the user from V | 01-01-2009 |
20090003626 | Dual Omnidirectional Microphone Array (DOMA) - A dual omnidirectional microphone array noise suppression is described. Compared to conventional arrays and algorithms, which seek to reduce noise by nulling out noise sources, the array of an embodiment is used to form two distinct virtual directional microphones which are configured to have very similar noise responses and very dissimilar speech responses. The only null formed is one used to remove the speech of the user from V | 01-01-2009 |
20090010449 | Microphone Array With Rear Venting - Microphone arrays (MAs) are described that position and vent microphones so that performance of a noise suppression system coupled to the microphone array is enhanced. The MA includes at least two physical microphones to receive acoustic signals. The physical microphones make use of a common rear vent (actual or virtual) that samples a common pressure source. The MA includes a physical directional microphone configuration and a virtual directional microphone configuration. By making the input to the rear vents of the microphones (actual or virtual) as similar as possible, the real-world filter to be modeled becomes much simpler to model using an adaptive filter. | 01-08-2009 |
20090010450 | Microphone Array With Rear Venting - Microphone arrays (MAs) are described that position and vent microphones so that performance of a noise suppression system coupled to the microphone array is enhanced. The MA includes at least two physical microphones to receive acoustic signals. The physical microphones make use of a common rear vent (actual or virtual) that samples a common pressure source. The MA includes a physical directional microphone configuration and a virtual directional microphone configuration. By making the input to the rear vents of the microphones (actual or virtual) as similar as possible, the real-world filter to be modeled becomes much simpler to model using an adaptive filter. | 01-08-2009 |
20090010451 | Microphone Array With Rear Venting - Microphone arrays (MAs) are described that position and vent microphones so that performance of a noise suppression system coupled to the microphone array is enhanced. The MA includes at least two physical microphones to receive acoustic signals. The physical microphones make use of a common rear vent (actual or virtual) that samples a common pressure source. The MA includes a physical directional microphone configuration and a virtual directional microphone configuration. By making the input to the rear vents of the microphones (actual or virtual) as similar as possible, the real-world filter to be modeled becomes much simpler to model using an adaptive filter. | 01-08-2009 |
20090022335 | Dual Adaptive Structure for Speech Enhancement - A clear, high quality voice signal with a high signal-to-noise ratio is achieved by use of an adaptive noise reduction scheme with two microphones in close proximity. The method includes the use of two omini directional microphones in a highly directional mode, and then applying an adaptive noise cancellation algorithm to reduce the noise. | 01-22-2009 |
20090034752 | CONSTRAINTED SWITCHED ADAPTIVE BEAMFORMING - An audio device, comprising a microphone array, a constrained switched adaptive beamformer with input coupled to said microphone array, said beamformer including (i) a first stage speech adaptive beamformer with first adaptive filters having a first adaptive step size, and (ii) a second stage noise adaptive beamformer with second adaptive filters having a second adaptive step size, and a single channel speech enhancer with input coupled to an output of said constrained switched adaptive beamformer. | 02-05-2009 |
20090034753 | Direction detection apparatus, direction detection method and direction detection program, and direction control apparatus, direction control method, and direction control program - A direction detection apparatus is disclosed. The direction detection apparatus includes a distribution obtainment section, an emphasis section, and a direction selection section. The distribution obtainment section obtains a distribution of intensities of sounds in a predetermined directional range. The emphasis section emphasizes sounds in the distribution of the intensities of the sounds obtained by the distribution obtainment section, wherein the emphasis section emphasizes the sounds in a second directional range which is a narrower directional range than the predetermined directional range and a center of the second directional range corresponds to a direction represented by selection information. The direction selection section decides a direction to be selected next based on the distribution of the intensities of the sounds which are output from the emphasis section and outputs the direction decided to be selected next as the selection information. | 02-05-2009 |
20090052686 | ELECTRONIC DEVICE WITH AN INTERNAL MICROPHONE ARRAY - An electronic device includes a front cover, a circuit board, a plurality of flexible boots, a plurality of microphones, and an abutting mechanism. The front cover includes a plurality of wall portions, a plurality of storage spaces encircled by the wall portions, and a plurality of acoustic openings connecting to the storage spaces. The flexible boots are disposed in the storage spaces. The microphones are mounted on the circuit board and disposed in the boots. The abutting mechanism pushes the circuit board toward the front cover to squeeze the flexible boot. | 02-26-2009 |
20090052687 | Method and apparatus for determining and indicating direction and type of sound - A method and apparatus for determining the direction of a sound source is disclosed. The method includes determining time differences of arrival of the sound at N locations and using the differences to determine the angular direction of the source. The apparatus indicates the angle of arrival and additionally indicates the type of the sound source. | 02-26-2009 |
20090052688 | REMOTE CONFERENCE APPARATUS AND SOUND EMITTING/COLLECTING APPARATUS - A speaker array and microphone arrays positioned on both sides of the speaker array are provided. A plurality of focal points each serving as a position of a talker are set in front of the microphone arrays respectively symmetrically with respect to a centerline of the speaker array, and a bundle of sound collecting beams is output toward the focal points. Difference values between sound collecting beams directed toward the focal points that are symmetrical with respect to the centerline are calculated to cancel sound components that detour from the speaker array to microphones. Then, it is estimated based on totals of squares of peak values of the difference values for a particular time period that the position of the talker is close to which one of the focal points, and the position of the talker is decided by comparing the totals of the squares of the peak values of the sound collecting beams directed to the focal points that are symmetrical mutually. | 02-26-2009 |
20090052689 | Deconvolution Methods and Systems for the Mapping of Acoustic Sources from Phased Microphone Arrays - Mapping coherent/incoherent acoustic sources as determined from a phased microphone array. A linear configuration of equations and unknowns are formed by accounting for a reciprocal influence of one or more cross-beamforming characteristics thereof at varying grid locations among the plurality of grid locations. An equation derived from the linear configuration of equations and unknowns can then be iteratively determined. The equation can be attained by the solution requirement of a constraint equivalent to the physical assumption that the coherent sources have only in phase coherence. The size of the problem may then be reduced using zoning methods. An optimized noise source distribution is then generated over an identified aeroacoustic source region associated with a phased microphone array (microphones arranged in an optimized grid pattern including a plurality of grid locations) in order to compile an output presentation thereof, thereby removing beamforming characteristics from the resulting output presentation. | 02-26-2009 |
20090060222 | Sound zoom method, medium, and apparatus - A sound zoom method, medium, and apparatus generating a signal in which a target sound is removed from sound signals input to a microphone-array by adjusting a null width that restricts a directivity sensitivity of the microphone array, and extracting a signal corresponding to the target sound from the sound signals by using the generated signal. Thus, a sound located at a predetermined position away from the microphone array can be selectively obtained so that a target sound is efficiently obtained. | 03-05-2009 |
20090074201 | METHOD AND APPARATUS FOR MICROPHONE MATCHING FOR WEARABLE DIRECTIONAL HEARING DEVICE USING WEARER'S OWN VOICE - Method and apparatus for microphone matching for wearable directional hearing assistance devices are provided. An embodiment includes a method for matching at least a first microphone to a second microphone, using a user's voice from the user's mouth. The user's voice is processed as received by at least one microphone to determine a frequency profile associated with voice of the user. Intervals are detected where the user is speaking using the frequency profile. Variations in microphone reception between the first microphone and the second microphone are adaptively canceled during the intervals and when the first microphone and second microphone are in relatively constant spatial position with respect to the user's mouth. | 03-19-2009 |
20090074202 | SYSTEM AND METHOD FOR LOCATING SOUND SOURCES - An exemplary method for locating sound sources is disclosed. The method includes the steps of: loading a sound source location program into a handheld device; activating the sound source location program; calculating a total voltage representing sound waves received by a microphone array via a waveform computation algorithm; calculating energy intensities of the total voltage according to the total voltage; and selecting a maximum energy intensity from the calculated energy intensities, and determining the location of the maximum energy intensity, the location of the maximum energy intensity is the location of the sound source. A related system is also disclosed. | 03-19-2009 |
20090086992 | MICROPHONE CIRCUIT AND CHARGE AMPLIFIER THEREOF - The invention provides a microphone circuit. In one embodiment, the microphone circuit comprises a microphone, a self-biased amplifier with a finite gain, and a feedback capacitor. The microphone coupled between a ground and a first node generates a first voltage at the first node according to sound pressure. The self-biased amplifier has a positive input terminal coupled to the ground and a negative input terminal coupled to the first node and amplifies the first voltage according to the finite gain to generate a second voltage at a second node. The feedback capacitor coupled between the first node and the second node feeds back the second voltage to the first node. The second voltage is then output to a following module subsequent to the microphone circuit. | 04-02-2009 |
20090086993 | Sound source direction detecting apparatus, sound source direction detecting method, and sound source direction detecting camera - Disclosed herein is a sound source direction detecting apparatus including: a plurality of microphones configured to collect sounds from a sound source in order to form an audio frame; a frequency decomposition section configured to decompose the audio frame into frequency components; an error range determination section configured to determine the effects of noises collected together with the sounds as an error range relative to phases; a power level dispersion section configured to disperse power levels of the sounds for each of the frequency components decomposed by the frequency decomposition section, on the basis of the error range determined by the error range determination section; a power level addition section configured to add the power levels dispersed by the power level dispersion section; and a sound source direction detection section configured to detect the direction of the sound source based on the phase at which is located the highest of the power levels added by the power level addition section. | 04-02-2009 |
20090103749 | Microphone Array Processor Based on Spatial Analysis - An array processing system improves the spatial selectivity by forming multiple steered beams and carrying out a spatial analysis of the acoustic scene. The analysis derives a time-frequency mask that, when applied to a reference look-direction beam (or other reference signal), enhances target sources and substantially improves rejection of interferers that are outside of the specified region. | 04-23-2009 |
20090110212 | Audio Transmission System and Communication Conference Device - In an audio transmission system, a control section of a communication conference device emits measurement sound waves from a loudspeaker array to a terminal unit and measures the time until a response is received, thereby detecting the position of the terminal unit. The control section sets directivity characteristics so that microphone sensitivity of the microphone array is brought to point to the position of the terminal unit, and sends the collected audio to another communication conference device. In a communication conference device on the reception side, the directivity characteristic of the loudspeaker array is set so that the received audio appears as if it was emitted from the position of the terminal unit on the transmission side. | 04-30-2009 |
20090129608 | Method for reducing interference powers and corresponding acoustic system - The object is to improve the action of a directional microphone in real acoustic environments. To do this, it is envisaged that the interference powers in a directional microphone with three microphones are reduced in that a first and a second microphone signal are adaptively filtered with respect to a first direction, with a direction-determining first parameter being adapted in such a way that the summation of interference powers is reduced. The second and a third microphone signal is adaptively filtered with respect to the first direction, with a direction-determining second parameter being adapted in such a way that the summation of interference powers is reduced. The two parameters are different from each other. This makes it possible, even in real environments, to suppress two interference sources from different directions with one second-order directional microphone. | 05-21-2009 |
20090129609 | Method and apparatus for acquiring multi-channel sound by using microphone array - Provided are a method and an apparatus for acquiring a multi-channel sound by using a microphone array. The method estimates positions of sound sources corresponding to sound source signals, which are mixed together, from the sound source signals input via a microphone array; and generates a multi-channel sound source signal by compensating for the sound source signals, based on differences between the estimated positions of the sound sources and a position of a virtual microphone array substituting for the microphone array. By doing so, the multi-channel sound having a stereoscopic effect can be acquired from a plurality of distant sound source signals which are input via the microphone array from a portable sound acquisition device. | 05-21-2009 |
20090136059 | MICROPHONE SYSTEM, SOUND INPUT APPARATUS AND METHOD FOR MANUFACTURING THE SAME - A microphone system, includes: a housing, adapted to be placed in a reference position relative to a sound source; a first microphone, configured to receive sound from the sound source at a first position within the housing; a second microphone, configured to receive sound from the sound source at a second position within the housing; and a differential signal generator, wherein: the first and second positions are arranged on a first line; and the first line perpendicularly intersects a second line that is extended from the sound source at a third position which is not between the first and second positions, and obliquely intersects a third line that is extended from the sound source at a fourth position which is between the first and second positions, when the housing is placed at the reference position. | 05-28-2009 |
20090141908 | Distance based sound source signal filtering method and apparatus - Provided is a sound source signal filtering method and apparatus. The sound source signal filtering method includes: generating two or more microphone output signals by combining sound source signals input through a plurality of microphones; calculating distances between the microphones and a sound source from which the sound source signals are emitted by using distance relationships according to frequencies of the sound source signals extracted from the generated microphone output signals; and filtering the sound source signals to obtain one or more sound source signals corresponding to a predetermined distance by using the calculated distances. Accordingly, it is possible to obtain only sound source signals emitted from a sound source at a particular distance from the microphone array among a plurality of sound source signals input through the microphone array. | 06-04-2009 |
20090147967 | CONFERENCE APPARATUS - Microphone arrays, which are formed by arranging a plurality of microphones, are provided on a front side and a rear side of a housing, respectively. A virtual focus is set for each of the microphone arrays in a direction opposite to a direction in which sound is picked-up sound signals picked up by the plurality of microphones are delayed such that distances to the virtual focus are the same, and the delayed sound signals are synthesized. Therefore, sound in a sound-pickup area of a predetermined angle on each of the front side and the rear side can be picked up at a high level, and even though there is a noise source in areas other than the sound-pickup area, noise from the noise source is not picked up. | 06-11-2009 |
20090190774 | ENHANCED BLIND SOURCE SEPARATION ALGORITHM FOR HIGHLY CORRELATED MIXTURES - An enhanced blind source separation technique is provided to improve separation of highly correlated signal mixtures. A beamforming algorithm is used to precondition correlated first and second input signals in order to avoid indeterminacy problems typically associated with blind source separation. The beamforming algorithm may apply spatial filters to the first signal and second signal in order to amplify signals from a first direction while attenuating signals from other directions. Such directionality may serve to amplify a desired speech signal in the first signal and attenuate the desired speech signal from the second signal. Blind source separation is then performed on the beamformer output signals to separate the desired speech signal and the ambient noise and reconstruct an estimate of the desired speech signal. To enhance the operation of the beamformer and/or blind source separation, calibration may be performed at one or more stages. | 07-30-2009 |
20090190775 | MICROPHONE ARRANGEMENT COMPRISING PRESSURE GRADIENT TRANSDUCERS - A microphone arrangement includes pressure gradient transducers that include a diaphragm. Each pressure gradient transducer has a first sound inlet opening that leads to a front portion of the diaphragm and a second sound inlet opening that leads to a back portion of the diaphragm. The directional characteristic of each pressure gradient transducer includes an omni portion, a figure-eight portion, and a direction of maximum sensitivity in a main direction. The acoustic centers of the pressure gradient transducers lie within an imaginary sphere having a radius corresponding to about double the largest dimension of the diaphragm. Projections of the main directions of the pressure gradient transducers form angles between about 110° and about 130° in a base plane. | 07-30-2009 |
20090190776 | SYNTHESIZING A MICROPHONE SIGNAL - A method synthesizes a microphone signal from a coincident microphone arrangement through multiple pressure gradient transducers. The pressure gradient transducers have directional characteristics that include an omni portion and a figure-eight portion. The direction of maximum sensitivity of the transducers lies within in a main direction. The method synthesizes a signal by forming a difference signal and a summed signal from the output of the two pressure gradient transducers. The difference and summed signals are converted into the frequency domain before the signals are spectrally subtracted. The method designates a representative phase to the magnitude of the spectrally subtracted signal. The phase corresponds to the phase of the summed signal. The signal and phase is then converted into the time domain. | 07-30-2009 |
20090190777 | MICROPHONE ARRANGEMENT HAVING MORE THAN ONE PRESSURE GRADIENT TRANSDUCER - A microphone arrangement includes multiple pressure gradient transducers having an acoustic center, a first sound inlet opening leading to a front of a diaphragm, and a second sound inlet opening leading the back of the diaphragm. A directional characteristic of the pressure gradient transducers includes an omni portion and a figure-eight portion. The pressure gradient transducers have a direction of maximum sensitivity in a main direction. Each main direction of the pressure gradient transducers is inclined. The acoustic center of a pressure transducer and the pressure gradient transducers are positioned within an imaginary sphere having a radius that corresponds to double the largest dimension of the diaphragm of one of the transducers. | 07-30-2009 |
20090214052 | SPEECH SEPARATION WITH MICROPHONE ARRAYS - A system that facilitates blind source separation in a distributed microphone meeting environment for improved teleconferencing. Input sensor (e.g., microphone) signals are transformed to the frequency-domain and independent component analysis is applied to compute estimates of frequency-domain processing matrices. Modified permutations of the processing matrices are obtained based upon a maximum magnitude based de-permutation scheme. Estimates of the plurality of source signals are provided based upon the modified frequency-domain processing matrices and input sensor signals. | 08-27-2009 |
20090214053 | POSITION DETERMINATION OF SOUND SOURCES - A microphone arrangement includes a database and multiple pressure gradient transducers having a diaphragm, a first sound inlet opening, and a second sound inlet opening. A directional characteristic of each of the pressure gradient transducers have a direction of maximum sensitivity in main directions. The main directions of the pressure gradient transducers are inclined. A pressure transducer has an acoustic center lying within an imaginary sphere with multiple acoustic centers of the pressure gradient transducer. The imaginary sphere has a radius corresponding to about double the largest dimension of the diaphragms of the pressure gradient transducers and the pressure transducer. The database retains representative signals of the multiple pressure gradient transducers and the pressure transducer. A processor accesses the database to determine a position of a sound source. | 08-27-2009 |
20090238377 | SPEECH ENHANCEMENT USING MULTIPLE MICROPHONES ON MULTIPLE DEVICES - Signal processing solutions take advantage of microphones located on different devices and improve the quality of transmitted voice signals in a communication system. With usage of various devices such as Bluetooth headsets, wired headsets and the like in conjunction with mobile handsets, multiple microphones located on different devices are exploited for improving performance and/or voice quality in a communication system. Audio signals are recorded by microphones on different devices and processed to produce various benefits, such as improved voice quality, background noise reduction, voice activity detection and the like. | 09-24-2009 |
20090238378 | Enhanced Immersive Soundscapes Production - An immersive audio-visual system (and a method) for creating an enhanced interactive and immersive audio-visual environment is disclosed. The immersive audio-visual environment enables participants to enjoy true interactive, immersive audio-visual reality experience in a variety of applications. The immersive audio-visual system comprises an immersive video system, an immersive audio system and an immersive audio-visual production system. The video system creates immersive stereoscopic videos that mix live videos, computer generated graphic images and human interactions with the system. The immersive audio system creates immersive sounds with each sound resource positioned correct with respect to the position of an associated participant in a video scene. The immersive audio-video production system produces an enhanced immersive audio and videos based on the generated immersive stereoscopic videos and immersive sounds. A variety of applications are enabled by the immersive audio-visual production including casino-type interactive gaming system and training system. | 09-24-2009 |
20090268925 | MICROPHONE ARRANGEMENT - A microphone arrangement includes multiple pressure gradient transducers having a diaphragm, a first sound inlet opening, and a second sound inlet opening. A directional characteristic of each of the pressure gradient transducers have a direction of maximum sensitivity in main directions. The main directions of the pressure gradient transducers are inclined. A pressure transducer has an acoustic center lying within an imaginary sphere with multiple acoustic centers of the pressure gradient transducer. The imaginary sphere has a radius corresponding to about double the largest dimension of the diaphragms of the pressure gradient transducers and the pressure transducer. | 10-29-2009 |
20090274318 | AUDIO CONFERENCE DEVICE - A audio conference device capable of detecting a talker's direction exactly and collecting a sound emitted from this direction at a high signal S/N ratio is provided. A detecting beam generating portion | 11-05-2009 |
20090279714 | APPARATUS AND METHOD FOR LOCALIZING SOUND SOURCE IN ROBOT - An apparatus and method for localizing a sound source in a robot are provided. The apparatus includes a microphone unit implemented by one or more microphones, which picks up a sound from a three-dimensional space. The apparatus also includes a sound source localizer for determining a position of the sound source in accordance with Time-Difference of Arrivals (TDOAs) and a highest power of the sound picked up by the microphone unit. Thus, the robot can rapidly and accurately localize the sound source in the three-dimensional space with minimum dead space, using a minimum number of microphones. | 11-12-2009 |
20090279715 | Method, medium, and apparatus for extracting target sound from mixed sound - A method, medium, and apparatus for extracting a target sound from a mixed sound. The method includes obtaining the mixed signal from a microphone array, generating a first signal which is emphasized and directed toward a target sound source, and a second signal which is suppressed and directed toward the target sound source, calculating a non-linear filter which is adaptive to at least one of an amplitude ratio of the first signal to the second signal in a time-frequency domain, frequencies of the first and second signals, and a ratio of an interference signal to the mixed signal, and filtering the first signal by the non-linear filter. | 11-12-2009 |
20090285409 | SOUND SOURCE LOCALIZATION DEVICE - Provided is a sound source localization device which can detect a source location of an extraction sound, including at least two microphones; an analysis unit ( | 11-19-2009 |
20090296957 | SOUND SYSTEM AND METHOD FOR CREATING A SOUND EVENT BASED ON A MODELED SOUND FIELD - A sound system and method for modeling a sound field generated by a sound source and creating a sound event based on the modeled sound field is disclosed. The system and method captures a sound field over an enclosing surface, models the sound field and enables reproduction of the modeled sound field. Explosion type acoustical radiation may be used. Further, the reproduced sound field may be modeled and compared to the original sound field model. | 12-03-2009 |
20090310797 | WIND NOISE REJECTION APPARATUS - An apparatus for reduction of wind noise comprised of an electro-acoustic transducer arrangement with at least two and preferably three omni-directional transducer elements. The exposed structure is covered with a thin layer of acoustic-resistive material. The electrical outputs of the elements are added together to provide an output signal with increased signal to noise ratio. | 12-17-2009 |
20090323980 | ARRAY MICROPHONE SYSTEM AND A METHOD THEREOF - An array microphone system of compensating phase drift of input signals and a method thereof. The microphone system comprises a speaker, first and second microphones, a delay estimator, and a memory. The speaker outputs an acoustic signal. The first and second microphones, spaced by a predetermined distance, receive the acoustic signal to generate first and second input signals. The delay estimator, coupled to the first and second microphones, computes a time difference between the first and second input signals. The memory, coupled to the delay estimator, stores the time difference. | 12-31-2009 |
20090323981 | Satellite Microphone Array For Video Conferencing - Speakers are identified based on sound origination detection through use of infrared detection of satellite microphones, estimation of distance between satellite microphones and base unit utilizing captured audio, and/or estimation of satellite microphone orientation utilizing captured audio. Multiple sound source localization results are combined to enhance sound source localization and/or active speaker detection accuracy. | 12-31-2009 |
20100008515 | MULTIPLE ACOUSTIC THREAT ASSESSMENT SYSTEM - A system is provided for locating and identifying an acoustic event. An acoustic sensor having a pair of concentric opposing microphones at a fixed distance on a microphone axis is used to measure an acoustic intensity, from which a vector incorporating the acoustic event is identified. A second acoustic sensor or movement of the first acoustic sensor is used to provide a second vector incorporating the acoustic event. Combination of the first and the second vector locates the acoustic event in space. A command unit in communication with the acoustic sensors can be used for combining the vectors as well as comparing a signal spectra of the acoustic event to stored identified spectra to provide an identification of acoustic event. | 01-14-2010 |
20100008516 | METHOD AND SYSTEM FOR POSITION DETECTION OF A SOUND SOURCE - A position detection method, system, and computer readable article of manufacture tangibly embodying computer readable instructions for executing the method for detecting the position of a sound source using at least two microphones. The method includes the steps of: emitting a reproduced sound from the sound source; observing the reproduced sound and an observed sound at the microphones; converting the reproduced sound and the observed sound into electrical signals; transforming the signals of the reproduced sound and of the observed sound into frequency spectra by a frequency spectrum transformer apparatus; calculating Crosspower Spectrum Phase (CSP) coefficients of the frequency spectra of the signals by a CSP coefficient calculator apparatus; and calculating distances between the position of the sound source and the positions of the microphones based on the calculated CSP coefficients by a distance calculating apparatus, thereby detecting the position of the sound source. | 01-14-2010 |
20100008517 | AUDIO SYSTEM BASED ON AT LEAST SECOND-ORDER EIGENBEAMS - A microphone array-based audio system that supports representations of auditory scenes using second-order (or higher) harmonic expansions based on the audio signals generated by the microphone array. In one embodiment, a plurality of audio sensors are mounted on the surface of an acoustically rigid sphere. The number and location of the audio sensors on the sphere are designed to enable the audio signals generated by those sensors to be decomposed into a set of eigenbeams having at least one eigenbeam of order two (or higher). Beamforming (e.g., steering, weighting, and summing) can then be applied to the resulting eigenbeam outputs to generate one or more channels of audio signals that can be utilized to accurately render an auditory scene. Alternative embodiments include using shapes other than spheres, using acoustically soft spheres and/or positioning audio sensors in two or more concentric patterns. | 01-14-2010 |
20100008518 | METHODS FOR PROCESSING AUDIO INPUT RECEIVED AT AN INPUT DEVICE - A method for processing an audio signal received through a microphone array coupled to an interfacing device is provided. The method is processing at least in part by a computing device that communicates with the interfacing device. The method includes receiving a signal at the microphone array and applying adaptive beam-forming to the signal to yield an enhanced source component of the signal. Also, an inverse beam-forming is applied to the signal to yield an enhanced noise component of the signal. The method combines the enhanced source component and the enhanced noise component to produce a noise reduced signal, where the noise reduced signal is a target voice signal. Then, monitoring an acoustic set-up associated with the audio signal as a background process using the adaptive beam-forming inverse beam-forming to track the target signal component, and periodically setting a calibration of the monitored acoustic set-up. The calibration implements blind source separation that uses second order statistics to separate the enhanced source component from the enhanced noise component, and the calibration remains fixed between the periodic setting. By executing this method, the target signal is able to freely move around relative to the microphone array of the interface device. | 01-14-2010 |
20100014689 | SYSTEMS AND METHODS FOR INTRA-ORAL BASED COMMUNICATIONS - Systems and methods are disclosed for capturing sound for communication by mounting one or more intra-oral microphones to capture sound; and mounting a mouth wearable communicator in the oral cavity to communicate sound with a remote unit. | 01-21-2010 |
20100014690 | Beamforming Pre-Processing for Speaker Localization - Embodiments of the present invention relate to methods, systems, and computer program products for signal processing. A first plurality of microphone signals is obtained by a first microphone array. A second plurality of microphone signals is obtained by a second microphone array different from the first microphone array. The first plurality of microphone signals is beamformed by a first beamformer comprising beamforming weights to obtain a first beamformed signal. The second plurality of microphone signals is beamformed by a second beamformer comprising the same beamforming weights as the first beamformer to obtain a second beamformed signal. The beamforming weights are adjusted such that the power density of echo components and/or noise components present in the first and second plurality of microphone signals is substantially reduced. | 01-21-2010 |
20100027808 | SOUND COLLECTION/REPRODUCTION METHOD AND DEVICE - To provide a sound collection system using a plurality of microphones arranged in the proximity to one another and having an excellent directivity for an arbitrary position in the sound field space. A plurality of control points are set around a plurality of sound collecting microphones. A desired response function matrix A(ω) and a transfer function matrix C(ω) between the control points and the respective microphones are measured. A control filter H arranged in a digital signal processing unit ( | 02-04-2010 |
20100027809 | METHOD FOR DIRECTING OPERATION OF MICROPHONE SYSTEM AND ELECTRONIC APPARATUS COMPRISING MICROPHONE SYSTEM - The invention provides a method for directing operation of a microphone system. In one embodiment, the microphone system comprises a plurality of component modules. First, a diagnostic test is performed to determine a diagnostic result indicating whether the component modules have failed the diagnostic test. Whether a plurality of required component modules corresponding to a current application mode for operating the microphone system have failed the diagnostic test is then determined according to the diagnostic result, wherein the application mode requires cooperation of the required component modules selected from the component modules of the microphone system. When some of the required component modules have failed the diagnostic test, the current application mode is changed to an altered application mode and the microphone system is directed to operate according to the altered application mode, wherein a plurality of second required component modules corresponding to the altered application mode are in good condition. When the required component modules are all in good condition, the microphone system is directed to operate according to the current application mode. | 02-04-2010 |
20100046770 | SYSTEMS, METHODS, AND APPARATUS FOR DETECTION OF UNCORRELATED COMPONENT - Detection of an uncorrelated component in a multi-channel acoustic signal is disclosed. In one example, the detection is based on a relation between (A) a difference in energy between two channels of the signal and (B) a threshold value that is based on an estimate of background energy of the acoustic signal. | 02-25-2010 |
20100054494 | MICROPHONE CIRCUIT - A microphone circuit includes a signal generating module, a filtering module, a transmitting module and a switch module. The signal generating module transforms audio signals into electronic signals. The filtering module is connected to the signal generating module to filter the electronic signals sent from the signal generating module. The transmitting module is connected to the filtering module to transmit the signals sent from the filtering module. The switch module is connected to the signal generating module to selectively regulate the microphone circuit to function as a differential microphone circuit or a single-ended microphone circuit. | 03-04-2010 |
20100054495 | Noise Mitigating Microphone System and Method - A microphone system has a base coupled with first and second microphone apparatuses. The first microphone apparatus is capable of producing a first output signal having a noise component, while the second microphone apparatus is capable of producing a second output signal. The first microphone apparatus may have a first back-side cavity and the second microphone may have a second back-side cavity. The first and second back-side cavities may be fluidly unconnected. The system also has combining logic operatively coupled with the first microphone apparatus and the second microphone apparatus. The combining logic uses the second output signal to remove at least a portion of the noise component from the first output signal. | 03-04-2010 |
20100092007 | Dynamic Switching of Microphone Inputs for Identification of a Direction of a Source of Speech Sounds - This disclosure describes techniques of automatically identifying a direction of a speech source relative to an array of directional microphones using audio streams from some or all of the directional microphones. Whether the direction of the speech source is identified using audio streams from some of the directional microphones or from all of the directional microphones depends on whether using audio streams from a subgroup of the directional microphones or using audio streams from all of the directional microphones is more likely to correctly identify the direction of the speech source. Switching between using audio streams from some of the directional microphones and using audio streams from all of the directional microphones may occur automatically to best identify the direction of the speech source. A display screen at a remote venue may then display images having angles of view that are centered generally in the direction of the speech source. | 04-15-2010 |
20100128892 | Stabilizing Directional Audio Input from a Moving Microphone Array - A device includes a microphone array fixed to the device. A signal processor produces an audio output using audio beamforming with input from the microphone array. The signal processor aims the beamforming in a selected direction. An orientation sensor—such as a compass, an accelerometer, or an inertial sensor—is coupled to the signal processor. The orientation sensor detects a change in the orientation of the microphone array and provides an orientation signal to the signal processor for adjusting the aim of the beamforming to maintain the selected direction. The device may include a camera that captures an image. An image processor may identify an audio source in the image and provide a signal adjusting the selected direction to follow the audio source. The image processor may receive the orientation signal and adjust the image for changes in the orientation of the camera before tracking movement of the audio source. | 05-27-2010 |
20100128893 | Communication system - There is provided a communication system having at least one headset ( | 05-27-2010 |
20100128894 | Acoustic Voice Activity Detection (AVAD) for Electronic Systems - Acoustic Voice Activity Detection (AVAD) methods and systems are described. The AVAD methods and systems, including corresponding algorithms or programs, use microphones to generate virtual directional microphones which have very similar noise responses and very dissimilar speech responses. The ratio of the energies of the virtual microphones is then calculated over a given window size and the ratio can then be used with a variety of methods to generate a VAD signal. The virtual microphones can be constructed using either an adaptive or a fixed filter. | 05-27-2010 |
20100128895 | SIGNAL PROCESSING UNIT, SIGNAL PROCESSING METHOD, AND RECORDING MEDIUM - A signal processing unit is provided. The signal processing unit includes an orthogonal transforming part including at least two sound input parts receiving input sound signals on a time axis, the orthogonal transforming part transforming two of the input sound signals into respective spectral signals on a frequency axis, a phase difference calculating part obtaining a phase difference between the two spectral signals on the frequency axis, and a filter part phasing, when the phase difference is within a given range, each component of a first one of the two spectral signals based on the phase difference at each frequency to calculate a phased spectral signal and combining the phased spectral signal and a second one of the two spectral signals to calculate a filtered spectral signal. | 05-27-2010 |
20100128896 | SOUND RECEIVING DEVICE, DIRECTIONAL CHARACTERISTIC DERIVING METHOD, DIRECTIONAL CHARACTERISTIC DERIVING APPARATUS AND COMPUTER PROGRAM - A sound receiving device | 05-27-2010 |
20100142725 | METHOD AND SYSTEM FOR SOUND MONITORING OVER A NETWORK - A mobile communication ( | 06-10-2010 |
20100158267 | Microphone Array Calibration Method and Apparatus - An apparatus for providing real-time calibration for two or more microphones. A calibrator for receiving a left microphone signal and a right microphone signal and generating phase difference data. A phase and amplitude correction system for receiving one of the left microphone signal or the right microphone signal the phase difference data and generating calibration data for a beamformer. The beamformer receiving the calibration data, the left microphone signal and the right microphone signal and generating a monaural beamformed signal. | 06-24-2010 |
20100158268 | TOROID MICROPHONE APPARATUS - A video teleconferencing directional microphone includes two microphone elements arranged coincidentally on a vertical axis. The two microphone elements are placed on a supporting surface so that a first microphone element is on the surface, and the second microphone elements are elevated above the supporting surface. The directional microphone also includes filters, an adder assembly, and an equalizer, which are used to shape the directivity pattern of the directional microphone into a toroid sensitivity pattern. The toroid sensitivity pattern increases sensitivity in the direction of a sound source of interest, while simultaneously reduces sensitivity to any sound waves generated by noise sources from certain elevation angles. | 06-24-2010 |
20100166212 | SOUND EMISSION AND COLLECTION DEVICE - It is possible to provide a sound emission and collection device having a compact configuration and being capable of suppressing a wraparound sound from a speaker to a microphone and improving the S/N ratio. In the sound emission and collection device, a plurality of speakers ( | 07-01-2010 |
20100177908 | ADAPTIVE BEAMFORMER USING A LOG DOMAIN OPTIMIZATION CRITERION - Described is a audio signal processing technology in which an adaptive beamformer processes input signals from microphones based on an estimate received from a pre-filter. The adaptive beamformer may compute its parameters (e.g., weights) for each frame based on the estimate, via a magnitude-domain objective function or log-magnitude-domain objective function. The pre-filter may include a time invariant beamformer and/or a non-linear spatial filter, and/or may include a spectral filter. The computed parameters may be adjusted based on a constraint, which may be selectively applied only at desired times. | 07-15-2010 |
20100177909 | BEAMFORMING SYSTEM COMPRISING A TRANSDUCER ASSEMBLY - A beamforming system (ASY) comprises a modular transducer assembly (MTA) composed of a plurality of transducer modules (TM | 07-15-2010 |
20100189279 | MICROPHONE ARRAY SIGNAL PROCESSING APPARATUS, MICROPHONE ARRAY SIGNAL PROCESSING METHOD, AND MICROPHONE ARRAY SYSTEM - A microphone array signal processing apparatus which is capable of picking up sound in a low frequency band even with a compact microphone array. The microphone array signal processing apparatus is comprised of delay devices ( | 07-29-2010 |
20100202628 | AUGMENTED ELLIPTICAL MICROPHONE ARRAY - In one embodiment, an audio system has a microphone array and a signal processing subsystem that processes audio signals generated by the microphone array to produce an output beampattem. The microphone array has (i) a plurality microphones arranged in a circular portion and (ii) a center microphone. The signal processing subsystem has (1) a decomposer that spatially decomposes the microphone audio signals to generate a plurality of eigenbeams and (2) a heamformer that generates the output beampattern as a weighted sum of the eigenbeams. By adding the center microphone, the audio system is able to provide some degree of control over the beamforming in the vertical direction as well as provide reduction of modal aliasin. | 08-12-2010 |
20100215189 | CEILING MICROPHONE ASSEMBLY - A video teleconferencing directional microphone has two surfaces joined with an angle of 90° relative to each other, a first omni directional microphone element arranged adjacent to the intersection between the two surfaces. The ceiling microphone assembly also includes a second omni directional microphone element arranged at a predetermined distance (d) from both surfaces. A subtractor subtracts the output of the first microphone element from the output of the second microphone element, and the output of the subtractor is equalized by an equalizer (H | 08-26-2010 |
20100226507 | Microphone Unit - A microphone unit comprises first and second microphones and a delay element. When sound is input to the first and second microphones, the delay element delays an output signal of the first microphone so as to detect the sound by a difference signal between the output signal of the first microphone and an output signal of the second microphone. The delay element delays the output signal of the first microphone so as to satisfy relation 0.76≦D/Δr≦2.0 where D is amount of delay for the output signal of the first microphone while Δr is distance between the first and second microphones. The relation D/Δr≦2.0 can reduce far-field noise, while the relation 0.76≦D/Δr can increase the detection sensitivity to sound emitted from a null point. | 09-09-2010 |
20100232620 | SOUND PROCESSING DEVICE, CORRECTING DEVICE, CORRECTING METHOD AND RECORDING MEDIUM - A sound processing device includes: a plurality of sound input units; a detecting unit for detecting a frequency component of each sound input to the plurality of sound signal unit, the each sound arriving from a direction approximately perpendicular to a line determined by arrangement positions of two sound input units among the plurality of sound input units; a correction coefficient unit for obtaining a correction coefficient for correcting a level of at least one of the sound signals generated from the input sounds by the two sound input units so as to match the levels of the sound signals with each other based on the sound of the detected frequency component; a correcting unit for correcting the level of at least one of the sound signals using the obtained correction coefficient; and a processing unit for performing a sound process based on the sound signal with the corrected level. | 09-16-2010 |
20100254543 | CONFERENCE MICROPHONE SYSTEM - A method and system for controlling selective audio output of captured sounds from an audience by means of a system comprising at least one microphone array located above or in front of said audience, and at least one camera. | 10-07-2010 |
20100266140 | VOICE INPUT/OUTPUT AUTOMATIC SWITCHING CIRCUIT USED IN HAND-HELD MICROPHONE WITH SPEAKER OF COMMUNICATION DEVICE SUCH AS TRANSCEIVER - An object of the invention is to provide a voice input/output automatic switching circuit used in a hand-held microphone with speaker of a communication device such as a transceiver. In the voice input/output automatic switching circuit, a circuit for detecting a handsfree speaker/microphone can be simply configured without requiring a particular detection terminal and moreover, a sufficient countermeasure can be taken against a malfunction caused upon its switching. The voice input/output automatic switching circuit includes a voice input automatic switching circuit constituted of: a current detecting resistor | 10-21-2010 |
20100272286 | Acoustic camera - An acoustic camera comprises a first sound pick-up device, a second sound pick-up device, and a switch. The switch is respectively connected to the first sound pick-up device and the second pick-up device and used to select the first sound pick-up device or the second sound pick-up device to reconstruct the sound field of the sound source of a detected object. The first sound pick-up device has a first microphone array, and the first microphone array is a near-field uniform microphone array. The second sound pick-up device has a second microphone array, and the second microphone array is a far-field non-uniform microphone array. | 10-28-2010 |
20100272287 | PATTERNED IMPLANTABLE ELECTRET MICROPHONE - An implantable microphone that includes a hermetically-sealed, enclosed volume and an electret member and back plate disposed with a space therebetween and capacitively coupleable to provide an output signal indicative of acoustic signals incident upon at least one of the electret member and back plate. At least one of the electret member and the back plate may include a plurality of laterally offset portions located in corresponding spatial relation to a plurality of laterally offset regions including the lateral extent of the space. The output signal may be at least one of weighted and weightable in relation to the plurality of laterally offset portions. The electret member may include the plurality of laterally offset portions, and the laterally offset portions may include at least one positively charged dielectric material portion and at least one negatively charged dielectric material portion. | 10-28-2010 |
20100303254 | AUDIO SOURCE DIRECTION DETECTING DEVICE - A sound source direction detector comprises FFT analysis sections ( | 12-02-2010 |
20100316231 | System and Method for Determining Vector Acoustic Intensity External to a Spherical Array of Transducers and an Acoustically Reflective Spherical Surface - A system and computer implemented method for determining and displaying vector acoustic intensity fields based on signals from a rigid spherical array of acoustic sensors within a volume external to the array. The method includes a propagator with a ratio of Green's functions for the location within the volume and for the spherical array radius, and a Tikhonov regularization filter that uses the Morozov discrepancy principle on the measured noise variance and Fourier coefficients of the measured partial pressures with respect to reference accelerometer or microphone measurements. | 12-16-2010 |
20100316232 | Spatial Audio for Audio Conferencing - Spatialized audio is generated for voice data received at a telecommunications device based on spatial audio information received with the voice data and based on a determined virtual position of the source of the voice data for producing spatialized audio signals. | 12-16-2010 |
20100316233 | METHOD OF OBJECT TRACKING IN 3D SPACE BASED ON PARTICLE FILTER USING ACOUSTIC SENSORS - There is provided a method of tracking an object in a three-dimensional (3-D) space by using particle filter-based acoustic sensors, the method comprising selecting two planes in the 3-D space; executing two-dimensional (2-D) particle filtering on the two selected planes, respectively; and associating results of the 2-D particle filtering on the respective planes. | 12-16-2010 |
20100322435 | Position Detecting System, Audio Device and Terminal Device Used in the Position Detecting System - A position detecting system is provided, which is capable of effectively preventing erroneous detection of audio to be measured. The position detecting system includes a terminal device that inputs an audio signal from an audio device and a microphone. The audio device sequentially inputs measurement audio signals that have been formed by two or more audio signals of different frequencies to a speaker and receives a notification signal, wherein the report signal indicates that the audio of the measurement audio signal has been collected from the terminal device. The audio device clocks a time t | 12-23-2010 |
20100322436 | ARRAY MICROPHONE SYSTEM INCLUDING OMNI-DIRECTIONAL MICROPHONES TO RECEIVE SOUND IN CONE-SHAPED BEAM - An array microphone system includes a first omni-directional microphone, a second omni-directional microphone, a gain control, and a beam former. The first omni-directional microphone faces a first direction. The second omni-directional microphone faces a second direction opposing the first direction. When receiving sound, the first omni-directional microphone and the second omni-directional microphone respectively generate a first signal and a second signal. The gain control amplifies the second signal to transform into a third signal, wherein strength of the third signal is equal to that of the first signal when the sound comes from the first direction. The beam former separates an in-beam sound signal and an out-beam sound signal from the first signal and the third signal. | 12-23-2010 |
20100329478 | HOUSING FOR MICROPHONE ARRAYS AND MULTI-SENSOR DEVICES FOR THEIR SIZE OPTIMIZATION - A sensor system being is located in an environment composed of a first medium, where waves propagate with a first phase velocity, the sensor system including at least one main enclosure and a sensor array with at least two sensors, said sensor array being arranged inside the main enclosure, wherein the space inside the main enclosure between the sensor array and the inner surface of the main enclosure is filled with a second medium, in which waves propagate with a second phase velocity, the second phase velocity being different from the first velocity. | 12-30-2010 |
20100329479 | SOUND SOURCE LOCALIZATION APPARATUS AND SOUND SOURCE LOCALIZATION METHOD - A sound source localization apparatus for localizing a sound source using an eigenvector, includes, a sound signal input unit inputting a sound signal, a correlation matrix calculation unit calculating a correlation matrix of the input sound signal, and an eigenvector calculation unit calculating an eigenvalue of the correlation matrix using the calculated correlation matrix, wherein the eigenvector calculation unit calculates the eigenvector using the correlation matrix of the input sound signal and one or more predetermined correlation matrices. | 12-30-2010 |
20110007911 | METHODS FOR LOCATING EITHER AT LEAST ONE SOUND GENERATING OBJECT OR A MICROPHONE USING AUDIO PULSES - In a first aspect, there is provided a method for locating a position of at least one sound generating object using at least one audio pulse, with the at least one audio pulse being detected by a plurality of stationary microphones located at a first position being spaced apart by a pre-determined distance. In a second aspect, there is provided a method for locating a position of a microphone using audio pulses emitted from a plurality of sound generating objects. The at least one audio pulse may preferably be in a form of a logarithmic swept sine (LSS) signal, as the LSS signal is detectable at both low volumes and amidst background noises. | 01-13-2011 |
20110019836 | SOUND PROCESSING APPARATUS - A sound emission and collection device includes a main housing and two sub-housings. In the main housing, a microphone array is provided. Microphone arrays are also provided in the sub-housings. Sound collection directions of the microphone arrays are outer directions which are opposite a side of the main housing. The sub-housings are rotatably connected to the main housing. The sound emission and collection device generates a plurality of collected sound beam signals MB | 01-27-2011 |
20110026730 | AUDIO PROCESSING APPARATUS AND METHOD - An audio processing apparatus is provided, comprising: a main microphone for receiving sounds from a source and noises from non-source sources and generating a main input; a reference microphone for receiving the sounds and the noises and generating a reference input; a short-time Fourier transformation (STFT) unit for applying short time Fourier transformation to convert the main input of a time domain signals into a main signal of a frequency domain and convert the reference input of the time domain signals into a reference signal of the frequency domain; a sensitivity calibrating unit for performing sensitivity calibration on the main signal and the reference signal and generating a main calibrated signal and a reference calibrated signal; and a voice active detector (VAD) for generating a voice active signal according to the main calibrated signal, the reference calibrated signal and a direction of arrival (DOA) signal. | 02-03-2011 |
20110026731 | MICROPHONE CIRCUIT AND METHOD FOR PREVENTING MICROPHONE CIRCUIT FROM GENERATING NOISE WHEN RESET - The invention provides a microphone circuit. In one embodiment, the microphone circuit comprises a transducer, a biasing resistor, a pre-amplifier, and a switch circuit. The transducer is coupled between a ground and a first node for converting a sound into a voltage signal output to the first node. The biasing resistor is coupled between the ground and the first node. The pre-amplifier is biased with a biasing voltage and coupled between the first node and a second node, and amplifies the voltage signal to obtain an output signal at the second node. The switch circuit is coupled between the first node and the ground, couples the first node to the ground when the microphone circuit is reset, and decouples the first node from the ground after a voltage status of the microphone circuit is stable, thus clamping a voltage of the first node to the ground to prevent generation of a popping noise when the microphone circuit is reset. | 02-03-2011 |
20110026732 | System for Detecting and Reducing Noise via a Microphone Array - A system for detecting noise in a signal received by a microphone array and a method for detecting noise in a signal received by a microphone array is disclosed. The system also provides for the reduction of noise in a signal received by a microphone array and a method for reducing noise in a signal received by a microphone array. The signal to noise ratio in handsfree systems may be improved, particularly in handsfree systems present in a vehicular environment. | 02-03-2011 |
20110033062 | ACOUSTIC VELOCITY MICROPHONE USING A BUOYANT OBJECT - Embodiments of a directional acoustic sensor or acoustic velocity microphone are disclosed that include a sensor frame structure, a support means, and a buoyant object. The buoyant object is suspended in the sensor frame structure using the support means. The buoyant object has a feature size smaller than a wavelength of the highest frequency of an acoustic wave in air. The buoyant object receives three-dimensional movement of the air excited by the acoustic wave. The three-dimensional movement that the buoyant object receives is detected using a detection means. A particle velocity of the acoustic wave is derived from the three-dimensional movement of the buoyant object using the detection means. The detection means can be an optical detection means, an electromagnetic detection means, or an electrostatic detection means. An acoustic image of the acoustic wave can be determined by distributing two or more directional acoustic sensors a multi-dimensional array. | 02-10-2011 |
20110033063 | SURROUND SOUND GENERATION FROM A MICROPHONE ARRAY - A signal from each of an array of microphones is analyzed. For at least one subset of microphone signals, a time difference is estimated, which characterizes the relative time delays between the signals in the subset. A direction is estimated from which microphone inputs arrive from one or more acoustic sources, based at least partially on the estimated time differences. The microphone signals are filtered in relation to at least one filter transfer function, related to one or more filters. A first filter transfer function component has a value related to a first spatial orientation of the arrival direction, and a second component has a value related to a spatial orientation that is substantially orthogonal in relation to the first. A third filter function may have a fixed value. A driving signal for at least two loudspeakers is computed based on the filtering. | 02-10-2011 |
20110038489 | SYSTEMS, METHODS, APPARATUS, AND COMPUTER-READABLE MEDIA FOR COHERENCE DETECTION - Based on phase differences between corresponding frequency components of different channels of a multichannel signal, a measure of directional coherency is calculated. Application of such a measure to voice activity detection and noise reduction are also disclosed. | 02-17-2011 |
20110051950 | Calibrating a Dual Omnidirectional Microphone Array (DOMA) - Systems and methods are described by which microphones comprising a mechanical filter can be accurately calibrated to each other in both amplitude and phase. | 03-03-2011 |
20110051951 | Calibrated Dual Omnidirectional Microphone Array (DOMA) - Systems and methods are described by which microphones comprising a mechanical filter can be accurately calibrated to each other in both amplitude and phase. | 03-03-2011 |
20110051952 | SOUND SOURCE IDENTIFYING AND MEASURING APPARATUS, SYSTEM AND METHOD - A sound source can be identified and measured for a long time period outdoors and indoors. A sound source identifying and measuring apparatus including a baffle provided with a frame and a weather-resistant screen for providing an aerial clearance is used for long-term indoor and outdoor measurement at a sound source measurement location to acquire sound source information in all the directions and associate the azimuth, elevation, sound pressure information and/or frequency characteristics or the like per elapsed time. A directional digital filter as well as identification parameters of a target sound source and untargeted sound source are used to identify the sound source more accurately for identification and measurement of the target sound source. Contribution of all of a plurality of sound sources to a sound pressure level is separated in terms of the coming direction for analysis. Thus, whether or not the sound source is a target sound is determined, and determination such as estimation of its sound source intensity and sound pressure level is made. | 03-03-2011 |
20110051953 | CALIBRATING MULTIPLE MICROPHONES - The specification and drawings present a new method, apparatus and software product for calibrating multiple microphones (e.g., a microphone array) to match their sensitivity using an ambient noise by creating and updating one or more calibration signal level difference histograms. | 03-03-2011 |
20110058683 | METHOD & APPARATUS FOR SELECTING A MICROPHONE IN A MICROPHONE ARRAY - A mobile robotic device includes a microphone array for detecting sound energy in its immediate environment. The sound energy received by each microphone in the microphone array is digitized, sampled and quantified. The quantified sound energy is used to calculate a sound energy difference factor between neighboring microphones in the array, the sound energy difference factors calculated over time are counted to be greater than or lesser than a nominal value and the counts are used to calculate a series of two-dimensional sound energy factors. The output of the microphone with the two highest calculated two-dimensional energy factors is then selected for processing and transmission over a network to be played at a far-end location. | 03-10-2011 |
20110069846 | AUDIO PROCESSING METHODS AND APPARATUSES UTILIZING THE SAME - An audio processing apparatus is provided. A microphone array includes microphone units. Amplifier modules each receives and amplifies an input signal from one microphone unit to generate amplified signals. A compensation module receives adjusted gains corresponding to the amplifier modules, obtains a gain difference between the adjusted gains, and adjusts one amplified signal according to the gain difference to obtain a compensated signal. | 03-24-2011 |
20110069847 | Sound collecting device, acoustic communication system, and computer-readable storage medium - There is provided a sound collecting device, including: an orientation direction forming section that forms an orientation direction of a microphone array; and a control section that, when a characteristic in a frequency band of a synthesized signal obtained by synthesizing the acoustic signals corresponds to a characteristic of an acoustic signal corresponding to a sound other than a target sound, controls the orientation direction forming section such that an orientation direction that is a direction that is different than an orientation direction of the microphone array at a present point in time is formed, and, when the characteristic in the frequency band of the synthesized signal does not correspond to a characteristic of an acoustic signal corresponding to a sound other than the target sound, controls the orientation direction forming section such that the orientation direction of the microphone array is maintained. | 03-24-2011 |
20110075857 | Apparatus for estimating sound source direction from correlation between spatial transfer functions of sound signals on separate channels - An apparatus estimates the direction of a sound source from signals plural microphones capture sound to produce. Data are stored on reverse characteristics of spatial transfer functions defined on sound transmitted from sound source positions to the respective microphones. To the signal produced by each microphone, applied are the reverse characteristics of the spatial transfer functions thus stored in connection with that microphone with respect to the sound source positions to thereby estimate a sound source signal on a sound source position associated with the sound captured. Between the sound source signals estimated on the sound source positions associated with the sounds captured by the microphones, coincidence or higher correlation is found on a sound source position to thereby produce information on at least the direction of the sound source thus found. | 03-31-2011 |
20110075858 | INFORMATION PROCESSING APPARATUS, INFORMATION PROCESSING METHOD, AND PROGRAM - There is provided an information processing apparatus including microphones, a parameter setting unit, and an audio signal processing unit. At least one pair of the microphones are provided, and the microphone picks up external audio to convert the external audio into an audio signal. The parameter setting unit sets a processing parameter specifying at least the sensitivity of the microphone according to at least an instruction from a user. Based on the processing parameter, the audio signal processing unit applies processing, including beamforming processing, to the audio signal input from the microphone. | 03-31-2011 |
20110075859 | APPARATUS FOR GAIN CALIBRATION OF A MICROPHONE ARRAY AND METHOD THEREOF - An apparatus and method for calibrating gain difference between microphones included in a microphone array are provided. In the gain calibrating apparatus, weights for each frequency component of the acoustic signals, which have been converted into the signals in the frequency domain are calculated. The weights are used to calibrate the acoustic signals such that the plurality of acoustic signals each have the same amplitude while the acoustic signals maintain their individual phase. The amplitudes of the acoustic signals are calibrated by use of the calculated weights. The gain calibrating apparatus calibrates gain in real time while calculating weights for frequency components of the frame of acoustic signals in real time. | 03-31-2011 |
20110085675 | Switchable Two-Element Directional Microphone System - A directional microphone system includes a first microphone that is disposed so as to receive sound energy originating from a first direction, convert the sound energy into a first voltage to be presented at a output. A second microphone is disposed to receive sound energy from a second direction different from the first direction and converts the sound energy into a second voltage presented at a second output. A first amplification circuit provides a first voltage gain at a third output. A second amplification circuit provides a second voltage gain at a fourth output and a predetermined time delay compared to the first amplification circuit. A controlled switch selectively switches a coupling of the first output between the first amplification circuit and the second amplification circuit and vice versa, and the coupling of the second output between the second amplification circuit and the first amplification circuit and vice versa. An output processing circuit is coupled to the first amplification circuit and the second amplification circuit and subtracts the fourth output from the third output such that a generally cardioid-shaped response is provided at its output. | 04-14-2011 |
20110096941 | SELF-STEERING DIRECTIONAL LOUDSPEAKERS AND A METHOD OF OPERATION THEREOF - A directional sound system, a method of transmitting sound to a spatial location determined by the gaze of a user and a directional communication system are disclosed. In one embodiment, the directional sound system includes: (1) a direction sensor configured to produce data for determining a direction in which attention of a user is directed, (2) a microphone configured to generate output signals indicative of sound received thereat, (3) loudspeakers configured to convert directed sound signals into directed sound and (4) an acoustic processor configured to be coupled to the direction sensor, the microphone, and the loudspeakers, the acoustic processor configured to convert the output signals to the directed sound signals and employ the loudspeakers to transmit the directed sound to a spatial location associated with the direction. | 04-28-2011 |
20110103611 | HEARING DEVICE AND METHOD FOR SUPPRESSING FEEDBACK WITH A DIRECTIONAL MICROPHONE - A hearing device and an associated method use an adaptive directional microphone for suppressing feedback. The hearing device includes an adaptation unit which sets the directional microphone so that a sound signal fed back from an earpiece of the hearing device to the directional microphone is attenuated. Acoustic feedback is advantageously attenuated or suppressed simply in an artifact-free manner. | 05-05-2011 |
20110103612 | Indoor Sound Receiving System and Indoor Sound Receiving Method - An indoor sound receiving system and an indoor sound receiving method are provided. The indoor sound receiving system comprises a microphone array, a path function database, a sound tracking unit, a path function selecting unit and a signal processing unit. The microphone array senses at least one primary sound source to output a plurality of microphones sensing signals. The path function database stores a plurality of sets of path functions. The sound tracking unit locates a primary sound source region according to a plurality of microphones sensing signals. The path function selecting unit selects a set of path functions corresponding to the primary sound source region as a set of primary sound source path functions from the path function database. The signal processing unit executes an audio enhancement process to output an enhanced speech signal according to the set of primary sound source path functions and the microphone sensing signals. | 05-05-2011 |
20110110531 | APPARATUS, METHOD AND COMPUTER PROGRAM FOR LOCALIZING A SOUND SOURCE - An apparatus for localizing a sound source includes at least two rotatably arranged microphones, a drive formed to set the microphones into rotation, and an evaluator. The evaluator is formed to receive microphone signals of the at least two microphones, while the at least two microphones are moving, and to obtain information on a direction from which sound arrives from the sound source or information on a position of the sound source, using the microphone signals obtained during the movement of the microphones. | 05-12-2011 |
20110129101 | Directional Microphone - A directional microphone system includes an ultrasonic emitter and receiver. The emitter directs a beam of ultrasound at the audio source with sufficient intensity that non-linear air effects cause non-linear interactions between the ultrasonic sound and the source's sonic sound. Ultrasonic frequency-mired sounds are thereby generated and these are received by the ultrasonic receiver. Signal-processing is carried out on the received signals to strip out the audio signals. The emitter and receiver may be co-located and the emitted beam may be focussed at the location of the audio source. The receiver may also be directional acid focussable. The directional microphone system may be very small and yet highly directional at sonic including low audible frequencies. | 06-02-2011 |
20110142252 | Source sound separator with spectrum analysis through linear combination and method therefor - In a source sound separator, first and second target sound predominant spectra are generated respectively by first and second processing operations for linear combination for emphasizing the target sound, using received sound signals of two microphones arrayed at a distance from each other. A target sound suppressed spectrum is generated by processing for linear combination for suppression of the target sound, using the two received sound signals. Further, a phase signal containing a larger amount of signal components of the target sound and exhibiting directivity in the direction of the target sound is generated by processing of linear combination, using the two received sound signals. The target sound and the interfering sound are separated from each other using the first and second target sound predominant spectra, the target sound suppressed spectrum, and the phase signal. | 06-16-2011 |
20110142253 | RECORDING/REPRODUCING APPARATUS - A recording/reproducing apparatus includes a plurality of unidirectional microphones and a plurality of direction indicator switches. The unidirectional microphones are each disposed in the periphery with a predetermined angle interval therebetween, whereas the direction indicator switches are able to indicate other directions other than a plurality of directions corresponding to the unidirectional microphones. In a normal reproducing mode, a plurality of audio signals which are picked up by the unidirectional microphones and subsequently recorded is read and reproduced in parallel. When any one of the direction indicator switches is operated, only the audio signal emitted in the designated direction is selectively read and reproduced. When another direction other than a plurality of directions corresponding to the unidirectional microphones is designated, audio signals picked up by two unidirectional microphones which are disposed to sandwich the designated direction is selectively read and reproduced. | 06-16-2011 |
20110158425 | MICROPHONE DIRECTIVITY CONTROL APPARATUS - A directivity control apparatus is capable of acquiring tilt information indicating a tilt angle of the directivity control apparatus; acquiring sound source direction information; storing mapping data indicating a relationship between the tilt angle and the direction; determining whether the sound information indicates a target sound; updating the mapping data based on the sound source direction information and the tilt information, if the sound information indicates the target sound; estimating a direction of sound responsive to the tilt information, based on the mapping data if the sound information doesn't indicate a target sound; and adjusting a directivity of a microphone based on the sound source direction information if the sound information indicates the target sound, or adjusting the directivity of the microphone based on the estimated direction if the sound information doesn't indicate the target sound. | 06-30-2011 |
20110158426 | SIGNAL PROCESSING APPARATUS, MICROPHONE ARRAY DEVICE, AND STORAGE MEDIUM STORING SIGNAL PROCESSING PROGRAM - A signal processing apparatus includes: two sound input units, an orthogonal transformer to transform two sound signals input from the two sound input units into respective spectral signals in a frequency domain, a phase difference calculator to calculate a phase difference between the spectral signals in the frequency domain, a range determiner to determine a coefficient responsive to a frequency in the phase difference as a function of frequency, and determine a suppression range related to a phase on a per frequency basis of the frequency responsive to the coefficient; and a filter to phase-shift a component of one of the spectral signals on a per frequency basis in order to generate a phase-shifted spectral signal when the phase difference at each frequency falls within the suppression range, synthesizing the phase-shifted spectral signal and the other of the spectral signals in order to generate a filtered spectral signal. | 06-30-2011 |
20110164760 | SOUND SOURCE TRACKING DEVICE - The sound source tracking device of the present invention comprises a plurality of differential microphones having bidirectionality, and a support member adapted to support the plurality of differential microphones such that the plurality of differential microphones are disposed in an array within a given plane. The plurality of differential microphones are supported on the support member such that their principal axes of directionality are approximately orthogonal to the given plane. | 07-07-2011 |
20110164761 | MICROPHONE ARRAY SYSTEM AND METHOD FOR SOUND ACQUISITION - A microphone array system ( | 07-07-2011 |
20110170705 | DIRECTIONAL MICROPHONE DEVICE - The directional microphone device according to the present invention solves a problem of increase in thermal noise (problem of decrease in sensitivity) that occurs at the time of directivity synthesis. The directional microphone device includes: a plurality of microphones which have directional and non-directional characteristics; a control unit which generates an output signal using signals outputted from each of the plurality of microphones; and an output unit which outputs the output signal generated by the control to unit. The control unit generates the output signal such that a nearly non-directional directivity and a high sensitivity are obtained in small amplitude range of the output signal, and a directivity and a low sensitivity are obtained in large amplitude range of the output signal. | 07-14-2011 |
20110176690 | INTEGRATED CIRCUIT DEVICE, VOICE INPUT DEVICE AND INFORMATION PROCESSING SYSTEM - There is provided an integrated circuit device having a wiring board | 07-21-2011 |
20110176691 | SOUND CONTROL APPARATUS, SOUND CONTROL METHOD, AND SOUND CONTROL PROGRAM - A sound control apparatus includes a direction accepting portion to accept designation of any one of a plurality of predetermined directions, a display control portion to allow a display portion to output a plurality of direction marks respectively indicating the plurality of directions, a plurality of microphones arranged at a distance away from each other, and a directivity control portion to control directivity of sounds respectively obtained by the plurality of microphones. The display control portion allows the display portion to display a direction mark corresponding to the direction accepted by the direction accepting portion in such a manner as to be enhanced as compared with any other direction mark. | 07-21-2011 |
20110200205 | SOUND PICKUP APPARATUS, PORTABLE COMMUNICATION APPARATUS, AND IMAGE PICKUP APPARATUS - A sound pickup apparatus includes: a microphone array including at least three microphones, wherein a first pair of microphones in which two of the at least three microphones are aligned on a first axis, and a second pair of microphones in which two of the at least three microphones are aligned on a second axis; a first null signal generator which outputs a first null signal based on a differential output of the first pair of microphones; a second null signal generator which outputs a second null signal based on a differential output of the second pair of microphones; and a combiner which generates a target signal based on the first null signal and the second null signal, the target signal having a directional characteristic in which the lowest sensitivity is formed in a direction to a line along which the first null surface meets the second null surface. | 08-18-2011 |
20110200206 | METHOD AND DEVICE FOR PHASE-SENSITIVE PROCESSING OF SOUND SIGNALS - A method and device for phase-sensitive processing of sound signals of at least one sound source may include arranging two microphones at a distance d from each other, capturing sound signals with both microphones, generating associated microphone signals, and processing the sound signals of the microphones. During a calibration mode, a calibration-position-specific, frequency-dependent phase difference vector φ | 08-18-2011 |
20110200207 | AUDIO APPARATUS - The invention provides an audio apparatus which hardly causes a directivity to be lowered even in a case where a plurality of unidirectional microphones, each having a directivity toward a center of a housing of the apparatus, are embedded in a recessed part provided on an upper surface of the housing. Microphones | 08-18-2011 |
20110206219 | Electronic device for receiving and transmitting audio signals - The present invention relates to an electronic device ( | 08-25-2011 |
20110222707 | SOUND SOURCE LOCALIZATION SYSTEM AND METHOD - A sound source localization system includes a plurality of microphones for receiving a signal as an input from a sound source; a time-difference extraction unit for decomposing the signal inputted through the plurality of microphones into time, frequency and amplitude using a sparse coding and then extracting a sparse interaural time difference (SITD) inputted through the plurality of microphones for each frequency; and a sound source localization unit for localizing the sound source using the SITDs. A sound source localization method includes receiving a signal as an input from a sound source; decomposing the signal into time, frequency and amplitude using a sparse coding; extracting an SITD for each frequency; and localizing the sound source using the SITDs. | 09-15-2011 |
20110222708 | BIOLOGY-INSPIRED MINIATURE SYSTEM AND METHOD FOR SENSING AND LOCALIZING ACOUSTIC SIGNALS - A system and method for sensing acoustic sounds is provided having at least one directional sensor, each directional sensor including at least two compliant membranes for moving in reaction to an excitation acoustic signal and at least one compliant bridge. Each bridge is coupled to at least a respective first and second membrane of the at least two membranes for moving in response to movement of the membranes it is coupled to for causing movement of the first membrane to be related to movement of the second membrane when either of the first and second membranes moves in response to excitation by the excitation signal. The directional sensor is controllably rotated to locate a source of the excitation signal, including determining a turning angle based on a linear relationship between the directionality information and sound source position described in experimentally calibrated data. | 09-15-2011 |
20110235821 | Variable directional microphone - There is provided a variable directional microphone including dynamic microphone units that is small in size and has good directional frequency response. In a variable directional microphone | 09-29-2011 |
20110235822 | APPARATUS AND METHOD FOR REDUCING REAR NOISE - An apparatus and method for removing noise are provided. The apparatus includes an acoustic signal input unit configured to comprise three or more microphones including a first microphone as a reference microphone, a second microphone disposed at a position asymmetrical to the first microphone, and a third microphone disposed at a position symmetrical to the first microphone, and an acoustic signal processing unit configured to remove rear noise using acoustic signals received from the first microphone, the second microphone, and the third microphone. | 09-29-2011 |
20110243347 | PIPE CALIBRATION DEVICE FOR CALIBRATION OF OMNIDIRECTIONAL MICROPHONES - Embodiments include a device comprising a pipe having at least one section that spans between a first end and a second end of the pipe. The pipe has a cylindrical cross-section. The device comprises a receptacle positioned in the pipe a first distance from the first end and a second distance from the second end. The receptacle receives an electronic device having microphones that are to be calibrated and secures the microphones a third distance inside an inside surface of the pipe. The device comprises an adapter connected to the first end. The adapter connects a loudspeaker to the pipe. The pipe controls an acoustic energy experienced by the plurality of microphones so that each microphone of the plurality of microphones receives equivalent acoustic energy. | 10-06-2011 |
20110243348 | PIPE CALIBRATION SYSTEM FOR OMNIDIRECTIONAL MICROPHONES - Embodiments include a system comprising a pipe that includes a first end and a second end. The pipe includes a plurality of sections coupled together. The system includes a loudspeaker that is a mouth simulator loudspeaker. The system includes an adapter that connects the loudspeaker to the first end. The system includes a receptacle positioned in the pipe a first distance from the first end and a second distance from the second end. The receptacle secures a plurality of microphones a third distance inside an inside surface of the pipe. | 10-06-2011 |
20110249830 | MICROPHONE UNIT - There is provided a microphone unit having a plurality of miniature microphones for respectively recording audio signals and a carrier unit. The miniature microphones can be arranged on a side of the carrier unit. | 10-13-2011 |
20110255709 | Audio control device and audio output device - An audio output device includes two digital microphone units that, upon receiving sound, convert the sound to PDM digital audio signals in which a state is represented by 1 or 0 in each predetermined period. The audio output device generates half-period digital audio signals, which are signals of a half period of the predetermined period, by using first digital audio signals and second digital audio signals that are the digital audio signals converted by the two digital microphones, where the states of the first digital audio signals are each reflected in one of two half periods corresponding to the predetermined period and states of the second audio signals are each reflected in the other half period. The audio output device then converts the half-period digital audio signals, which are generated by the generator, to analog audio signals and outputs the analog audio signals. | 10-20-2011 |
20110261973 | APPARATUS AND METHOD FOR REPRODUCING A SOUND FIELD WITH A LOUDSPEAKER ARRAY CONTROLLED VIA A CONTROL VOLUME - Method and, apparatus for implementing the method, the method comprising determining control signal data for an array of loudspeakers, the control signal data being such as to control the loudspeakers to produce a desired sound field associated with an audio signal, the method comprises determining control signal data for different frequency components of the desired sound field in respect of respective different positions in a listening volume of the loudspeaker array, wherein determination of the control signal data comprises sampling the desired sound field at the surface of a control volume (V). | 10-27-2011 |
20110268292 | Apparatus - Apparatus including: an acoustic transducer, and a sound channel coupled to the acoustic transducer, the sound channel including an element having a shape that is electrically controllable, wherein the shape of the element is electrically controllable to change the acoustic properties of the sound channel. | 11-03-2011 |
20110274289 | SENSOR ARRAY BEAMFORMER POST-PROCESSOR - A novel beamforming post-processor technique with enhanced noise suppression capability. The present beamforming post-processor technique is a non-linear post-processing technique for sensor arrays (e.g., microphone arrays) which improves the directivity and signal separation capabilities. The technique works in so-called instantaneous direction of arrival space, estimates the probability for sound coming from a given incident angle or look-up direction and applies a time-varying, gain based, spatio-temporal filter for suppressing sounds coming from directions other than the sound source direction, resulting in minimal artifacts and musical noise. | 11-10-2011 |
20110286609 | MULTIPLE MICROPHONE BASED DIRECTIONAL SOUND FILTER - A system and method for use in filtering of an acoustic signal are provided for producing an output signal of attenuated amount of diffuse sound in accordance with predetermined parameters of desired output directional response and required attenuation of diffuse sound. The system includes a filtration module and a filter generation module including a directional analysis module and filter construction module. | 11-24-2011 |
20110286610 | SURFACE MICROMACHINED DIFFERENTIAL MICROPHONE - A differential microphone having a perimeter slit formed around the microphone diaphragm that replaces the backside hole previously required in conventional silicon, micromachined microphones. The differential microphone is formed using silicon fabrication techniques applied only to a single, front face of a silicon wafer. The backside holes of prior art microphones typically require that a secondary machining operation be performed on the rear surface of the silicon wafer during fabrication. This secondary operation adds complexity and cost to the micromachined microphones so fabricated. Comb fingers forming a portion of a capacitive arrangement may be fabricated as part of the differential microphone diaphragm. | 11-24-2011 |
20110293107 | SOUND SIGNAL PROCESSING APPARATUS AND SOUND SIGNAL PROCESSING METHOD - A sound signal processing apparatus includes a sound source direction determination unit and a filter processing unit. The sound source direction determination unit determines sound source directions with respect to sound signals of a plurality of channels for respective first to n-th bands. The filter processing unit includes first to n-th filters which are connected in series and configured to boost or attenuate the sound signals with respect to the first to n-th bands. The respective first to n-th filters perform boosting or attenuation based on the sound source directions of the first to n-th bands which are determined by the sound source direction determination unit. | 12-01-2011 |
20110293108 | SYSTEM AND METHOD FOR PRODUCING A DIRECTIONAL OUTPUT SIGNAL - A system and method of producing a directional output signal is described including the steps of: detecting sounds at the left and rights sides of a person's head to produce left and right signals; determining the similarity of the signals; modifying the signals based on their similarity; and combining the modified left and right signals to produce an output signal. | 12-01-2011 |
20110299701 | MINIATURE MICRO-ELECTROMECHANICAL SYSTEM (MEMS) BASED DIRECTIONAL SOUND SENSOR - A micro-electromechanical (MEMS) based directional sound sensor includes a two sensor wings attached to a surrounding support structure by two legs. The support structure is hollow beneath the sensor wings allowing the sensor wings to vibrate in response to sound excitation. In one embodiment, interdigitated comb finger capacitors attached on the sensor wing edges and the support structure enable an electrostatic (capacitive) readout related to the vibrations of the sensor which allows determination of the sound direction. | 12-08-2011 |
20110299702 | APPARATUS, METHOD AND COMPUTER PROGRAM FOR PROVIDING A SET OF SPATIAL CUES ON THE BASIS OF A MICROPHONE SIGNAL AND APPARATUS FOR PROVIDING A TWO-CHANNEL AUDIO SIGNAL AND A SET OF SPATIAL CUES - An apparatus for providing a set of spatial cues associated with an upmix audio signal having more than two channels on the basis of a two-channel microphone signal has a signal analyzer and a spatial side information generator. The signal analyzer is configured to obtain a component energy information and a direction information on the basis of the two-channel microphone signal, such that the component energy information describes estimates of energies of a direct sound component of the two-channel microphone signal and of a diffuse sound component of the two-channel microphone signal, and such that the directional information describes an estimate of a direction from which the direct sound component of the two-channel microphone signal originates. The spatial side information generator is configured to map the component energy information and the direction information onto a spatial cue information describing the set of spatial cues associated with an upmix audio signal having more than two channels. | 12-08-2011 |
20120014535 | SOUND COLLECTION DEVICE - The invention provides a sound collection device having little error in a desired directivity. The sound collection device includes a unidirectional microphone | 01-19-2012 |
20120063613 | RECORDING APPARATUS, RECORDING CONDITION SETTING METHOD, AND NON-TRANSITORY COMPUTER-READABLE RECORDING MEDIUM ENCODED WITH RECORDING CONDITION SETTING PROGRAM - A recording apparatus includes: a plurality of microphones having directivity to output collected sound; a switch portion to switch a direction of directivity of each of the plurality of microphones to one of a plurality of predetermined direction patterns; a detection portion to detect a direction pattern switched by the switch portion among the plurality of direction patterns; a recording portion to execute plural kinds of processing on sound collected by the plurality of microphones and to record the processed sound; a setting portion to set parameters to be used by the recording portion to execute the plural kinds of processing; and a storage portion to store the parameters to be used by the recording portion to execute the plural kinds of processing, separately for each of the plural kinds of processing, in association with each of the plurality of direction patterns. When a direction pattern switched by the switch portion is detected by the detection portion, the setting portion sets the parameters to be used to execute plural kinds of processing that are associated with the detected direction pattern. | 03-15-2012 |
20120070015 | APPARATUS AND METHOD FOR ENHANCING AUDIO QUALITY USING NON-UNIFORM CONFIGURATION OF MICROPHONES - An audio quality enhancing apparatus and method is provided in which a microphone array has a non-uniform configuration and thus a beam pattern of a desired direction is obtained in a wide range of frequencies including higher frequency bands and lower frequency bands even when the microphone array is relatively small. The audio quality enhancing apparatus includes at least three microphones which are disposed in a non-uniform configuration, a frequency conversion unit configured to transform acoustic signals input from the at least three microphones to acoustic signals of frequency domain; a band division and merging unit configured to divide frequencies of the transformed acoustic signals into bands based on intervals between the at least three microphones and to merge the acoustic signals in the frequency domain into signals of two channels based on the divided frequency bands; and a two channel beamforming unit configured to reduce noise of signals including input from a direction other than the direction of a target sound by performing beamforming on the signals of the two channels and to output the noise-reduced signals. | 03-22-2012 |
20120082322 | SOUND SCENE MANIPULATION - An audio-processing device having an audio input, for receiving audio signals, each audio signal having a mixture of components, each corresponding to a sound source, and a control input, for receiving, for each sound source, a desired gain factor associated with the source, by which it is desired to amplify the corresponding component, and an auxiliary signal generator, for generating at least one auxiliary signal from the audio signals, and with a different mixture of components as compared with a reference audio signal; and a scaling coefficient calculator, for calculating scaling coefficients based upon the desired gain factors and upon parameters of the different mixture, each scaling coefficient associated with one of the auxiliary signal and optionally the reference audio signal, and an audio synthesis unit, for synthesizing an output audio signal by applying scaling coefficients to the auxiliary signal and optionally the reference audio signal and combining the results. | 04-05-2012 |
20120087512 | DISTRIBUTED SIGNAL PROCESSING SYSTEMS AND METHODS - Systems and methods for parallel and distributed processing of audio signals produced by a microphone array are described. In one aspect, a distributed signal processing system includes an array of microphones and an array of processors. Each processor is connected to one of the microphones and is connected to at least two other processors, enabling communication between adjacent connected processors. The system also includes a computing device connected to each of the processors. Each microphone detects a sound and generates an audio signal, and each processor is configured to receive and process the audio signal sent from a connected microphone and audio signals sent from at least one of the adjacent processors to produce a data stream that is sent to the computing device. | 04-12-2012 |
20120087513 | MICROPHONE UNIT AND SOUND COLLECTING DEVICE - To provide a microphone unit capable of acquiring a target sound with high accuracy. A microphone unit in accordance with an exemplary embodiment of the present invention includes a plurality of microphones, a microphone substrate on which the plurality of microphones are mounted, and a vibration observation device disposed at roughly a center of gravity of a shape that is formed by connecting centers of certain adjacent microphones among the plurality of microphones. | 04-12-2012 |
20120093336 | SYSTEMS AND METHODS FOR PERFORMING SOUND SOURCE LOCALIZATION - Systems and methods for performing sound source localization are provided. In one aspect, a method for locating a sound source using a computing device subdivides a space into subregions. The method then computes a sound source power for each of subregions and determines which of the sound source energies is the largest. When the volume of the subregion is less than a threshold volume, the method outputs the subregion having the largest sound source power. Otherwise, the stages of partitioning, computing, and determining the subregion having the largest sound source power is repeated. | 04-19-2012 |
20120093337 | Microphone Array - Embodiments of the invention improve upon the prior art array by having more carefully defined directivity functions designed to meet two criteria, being firstly to minimise cross-talk between non-adjacent microphones in the array, and secondly to design the array response such that it approximates stereophonic panning curves that have been shown to provided for good auditory localisation. One embodiment therefore provides a microphone array, comprising N microphones, wherein N is greater than or equal to 3. The microphones are substantially equiangularly arranged over a circular arc subtending an angle ε, wherein ε is less than or equal to 2π, with the directional axes of the N microphones facing substantially radially outwards. Each of the N microphones have a substantially common directivity function Γ(θ) defining the directional response of the microphone, wherein θ=0 is the directional axis, and the directivity function Γ(θ) is arranged such that a sound source in acoustical free field is effectively captured by no more than two consecutive microphones in the array. By arranging the directivity function in this manner crosstalk between non-adjacent microphones can be minimised, which has been shown to improve auditory localisation performance. | 04-19-2012 |
20120093338 | SYSTEM AND METHOD FOR SPATIAL NOISE SUPPRESSION BASED ON PHASE INFORMATION - Disclosed herein are systems, methods, and non-transitory computer-readable storage media for suppressing spatial noise based on phase information. The method transforms audio signals to frequency-domain data and identifies time-frequency points that have a parameter (e.g., signal-to-noise ratio) above a threshold. Based on these points, unwanted signals can be attenuated the desired audio source can be isolated. The method can work on a microphone array that includes two microphones or more. | 04-19-2012 |
20120093339 | 3D SOUNDSCAPING - A system and method for tracking and tracing motions of multiple incoherent sound sources and for visualizing the resultant overall sound pressure distribution in 3D space in real time are developed. This new system needs only four microphones (although more could be used) that can be mounted at any position so long as they are not placed on the same plane. A sample configuration is to mount three microphones on the y, z plane, while the 4th microphone on a plane perpendicular to the y, z plane. A processor receives signals from the microphones based on the signals received from noise sources in unknown locations, and the processor determines the locations of these sources and visualizes the resultant sound field in 3D space in real time. This system works for broadband, narrowband, tonal sound signals under transient and stationary conditions. | 04-19-2012 |
20120093340 | VARIABLE DIRECTIONAL MICROPHONE UNIT AND VARIABLE DIRECTIONAL MICROPHONE - A variable directional microphone unit includes, a pair of microphone elements disposed back to back, output signal systems of the microphone elements connected to a hot-side terminal and a cold-side terminal of a balanced output respectively, an inverting amplifier connected to one output signal system of the microphone elements, an input resistance and a feedback resistance of the inverting amplifier at least any one of which is divided, and a switching device switching a signal retrieving point by arbitrarily selecting each divide of at least one of the input resistance or the feedback resistance. The switching device switches one output of the balanced output to enable directivity of the balanced output signal to vary. A circuit for switching the directivity does not become a load or a noise source. | 04-19-2012 |
20120099739 | ESTIMATION OF SYNTHETIC AUDIO PROTOTYPES - An approach to forming output signals both permits flexible and temporally and/or frequency local processing of input signals while limiting or mitigating artifacts in such output signals. Generally, the approach involves first synthesizing prototype signals for the output signals, or equivalently characterizing such prototypes, for example, according to their statistical characteristics, and then forming the output signals as estimates of the prototype signals, for example, as weighted combinations of the input signals. | 04-26-2012 |
20120106753 | DIGITAL DUAL MICROPHONE MODULE WITH INTELLIGENT CROSS FADING - A method of operating a microphone system includes providing first and second microphones associated with a same human speaker. An analog ambient noise signal is received from the first microphone. An analog speech signal is received from the second microphone. The analog ambient noise signal is converted into a digital ambient noise signal. The analog speech signal is converted into a digital speech signal. Digital noise cancellation is performed on the digital speech signal dependent upon the digital ambient noise signal. The digital noise cancellation is performed by digital circuitry. The noise canceled digital speech signal is inputted into an intercom system. A low power condition of the microphone system and/or a failure of the digital circuitry is sensed. In response to the sensing step, an analog-based intercom signal is inputted into the intercom system. The analog-based intercom signal is dependent on the analog speech signal and substantially independent of the analog ambient noise signal. The analog-based intercom signal is inputted into the intercom system without noise cancellation having been performed on the analog-based intercom signal. | 05-03-2012 |
20120106754 | TRANSITIONING MULTIPLE MICROPHONES FROM A FIRST MODE TO A SECOND MODE - An apparatus includes multiple microphones and a controller. The controller is coupled to receive a signal from each of the multiple microphones. The controller is configured to control a transition of the multiple microphones from an active mode to a dormant mode. When the multiple microphones are in the active mode, the controller is configured to perform signal processing responsive to signals received from at least two of the multiple microphones. When the multiple microphones are in the dormant mode, the controller is configured to select a microphone of the multiple microphones and to perform signal processing corresponding to the selected microphone while suspending signal processing corresponding to unselected microphones. | 05-03-2012 |
20120106755 | HANDHELD ELECTRONIC DEVICE WITH MICROPHONE ARRAY - A handheld electronic device includes a body and a microphone array. The body includes a side and a recess formed on the side. The microphone array includes a first microphone and a second microphone. Either the first and second microphones are disposed in the recess, or the first microphone is disposed in the recess while the second microphone is disposed outside the recess. | 05-03-2012 |
20120114137 | Acoustic Control Apparatus and Acoustic Control Method - Disclosed herein is an acoustic control apparatus including: a speaker-position computation section configured to find the position of each of a plurality of speakers located in a speaker layout space on the basis of a position computed as the microphone position in the speaker layout space based on a taken image of at least any of the microphone and an object placed at a location close to the microphone position, and a result of sound collection to collect a signal sound each generated by one of the speakers; and an acoustic control section configured to control a sound generated by each of the speakers by computing a user position in the speaker layout space based on a taken image of the user, computing the distance between the user position and the position of each of the speakers, and controlling sounds generated by the speakers according to the computed distances. | 05-10-2012 |
20120114138 | SOUND SOURCE SIGNAL PROCESSING APPARATUS AND METHOD - A sound source signal processing apparatus including a first sound source detection unit having at least one microphone to detect a sound source signal, a second sound source detection unit having at least one microphone to detect the sound source signal, the second sound source detection unit being spaced apart from the first sound source detection unit, and a beamforming unit to beamform the sound source signal detected by the first sound source detection unit and the second sound source detection unit. At least one microphone is further provided in addition to the microphone array, and position information of the microphones and sound source information are used, thereby improving beamforming performance of the sound source signal. Also, the number and size of microphone arrays is reduced through further provision of the at least one microphone, thereby improving spatial utilization. | 05-10-2012 |
20120128174 | Converting multi-microphone captured signals to shifted signals useful for binaural signal processing and use thereof - A method includes, for each of a number of subbands of a frequency range and for at least first and second frequency-domain signals that are frequency-domain representations of corresponding first and second audio signals: determining a time delay of the first frequency-domain signal that removes a time difference between the first and second frequency-domain signals in the subband. The method includes forming a first resultant signal including, for each of the number of subbands, a sum of one of the first or second frequency-domain signals shifted by the time delay and of the other of the first or second frequency-domain signals; and forming a second resultant signal including, for each of the number of subbands, a difference between the shifted one of the first or second frequency-domain signals and the other of the first or second frequency-domain signals. Apparatus and program products are also disclosed. | 05-24-2012 |
20120128175 | 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. | 05-24-2012 |
20120128176 | SPATIAL NOISE SUPPRESSION FOR A MICROPHONE ARRAY - A noise reduction system and a method of noise reduction includes utilizing an array of microphones to receive sound signals from stationary sound sources and a user that is speaking. Positions of the stationary sound sources relative to the array of microphones are estimated using sound signals emitted from the sound sources at an earlier time. Noise is suppressed in an audio signal based at least in part on the estimated positions of the stationary sound sources. A position of the user relative to the array of microphones can also be estimated | 05-24-2012 |
20120134507 | Methods, Systems, and Products for Voice Control - Methods, systems, and computer program products provide voice control of electronic devices. Speech and a beacon signal are received. A directional microphone is aligned to a source of the beacon signal. A voice command in the speech is received and executed. | 05-31-2012 |
20120140946 | Wind Noise Mitigation - A method of compensating for noise in a receiver having a first receiver unit and a second receiver unit, the method includes receiving a first transmission at the first receiver unit, the first transmission having a first signal component and a first noise component; receiving a second transmission at the second receive unit, the second transmission having a second signal component and a second noise component; determining whether the first noise component and the second noise component are incoherent and; only if it is determined that the first and second noise components are incoherent, processing the first and second transmissions in a first processing path, wherein the first processing path is configured to compensate for incoherent noise. | 06-07-2012 |
20120140947 | Apparatus and method to localize multiple sound sources - An apparatus and method to localize multiple sound sources is provided. Virtual microphone signals are generated based on actual microphone signals from a microphone array including a plurality of microphones, which are arranged at intervals that may minimize space aliasing at a given sampling frequency, and sound source directions are tracked using the actual microphone signals and the virtual microphone signals. Thus, without increasing the aperture length of the microphone array, it is possible to achieve almost the same resolution as when a microphone array having a relatively long length is used. | 06-07-2012 |
20120140948 | DIRECTIONAL MICROPHONE DEVICE AND DIRECTIVITY CONTROL METHOD - A directional microphone apparatus and directivity control method that corrects a level difference and a phase difference generated in a low band in a plurality of non-directional microphone units, improve the directivity, and reduce the size are provided. Level difference calculation section ( | 06-07-2012 |
20120148067 | WIND NOISE DETECTION METHOD AND SYSTEM - The present invention relates to a multi-microphone system and method adapted to determine phase angle differences between a first microphone and a second microphone signal to detect presence of wind noise. | 06-14-2012 |
20120163622 | NOISE DETECTION AND REDUCTION IN AUDIO DEVICES - Methods and apparatuses for detection and reduction of wind noise in audio devices are disclosed. In an embodiment, a method includes acquiring and transforming the audio signals. Correlations from the transformed audio signals are computed. A cross correlation index is compared to a predetermined value to determine if a wind noise spectral content is present. In another embodiment, an apparatus includes an audio processing unit to receive non-decomposed audio signals, and an audio decomposition unit to receive the non-decomposed audio signals and to generate decomposed audio signals. A wind noise spectrum estimation unit receives non-decomposed audio signals and decomposed audio signals and identifies wind noise spectral components in at least one of the non-decomposed and decomposed audio signals. A wind noise spectrum reduction unit receives the wind noise spectral components and removes the wind noise spectral components from at least one of the non-decomposed and the decomposed audio signals. | 06-28-2012 |
20120163623 | WIDEBAND NOISE REDUCTION SYSTEM AND A METHOD THEREOF - Systems and methods improve audio signals and include means and methods of reducing stochastic noise in wideband audio signals. Multiple microphones may acquire near and far end audio signals, the audio signals may undergo transformations via a general or specialized digital signal processor. | 06-28-2012 |
20120163624 | Directional sound source filtering apparatus using microphone array and control method thereof - A directional sound source filtering apparatus using a microphone array and a control method thereof are provided. The directional sound source filtering apparatus using a microphone array includes an image detector to detect images in a destination area, a sound collector located by the microphone array in which microphones are arranged to detect sound sources together with the images detected by the image detector. The apparatus includes a controller to precalculate time delay values of sound sources within the images in order to extract sound sources within the image from the sound sources detected by the sound collector, and perform beamforming through the calculated time delay values. Sound source signals only within images may be selectively amplified using beamformers. | 06-28-2012 |
20120163625 | METHOD OF CONTROLLING AUDIO RECORDING AND ELECTRONIC DEVICE - A method of controlling audio recording using an electronic device and an electronic device are described. The electronic device comprises a microphone arrangement having a directivity pattern. A target direction relative to the electronic device is automatically determined in response to sensor data representing at least a portion of an area surrounding the electronic device. The microphone arrangement is automatically controlled in response to the determined target direction to adjust an angular orientation of the directivity pattern relative to the electronic device. | 06-28-2012 |
20120163626 | ACTIVE SOUND REDUCTION SYSTEM AND METHOD - The present invention refers to an active sound reduction system and method for attenuation of sound emitted by a primary sound source, especially for attenuation of snoring sounds emitted by a human being. This system comprises a primary sound source, at least one speaker as a secondary sound source for producing an attenuating sound to be superposed with the sound emitted by said primary sound source, a reference microphone for receiving sound from said primary sound source, and at least one error microphone being allocated to each speaker to form a speaker/microphone pair. The at least one error microphone is provided as a directional microphone pointing at its allocated speaker to receive residual sound resulting from the superposition of the sounds from the primary sound source and the corresponding speaker. The error microphone and speaker of at least one speaker/microphone pair and the primary sound source are arranged substantially collinear. A control unit is provided to receive an output reference signal of the reference microphone representing the sound received by the reference microphone and an output error signal of the at least one error microphone representing the sound received by the at least one error microphone and to calculate a control signal for the speaker from the output reference signal and the output error signal. | 06-28-2012 |
20120177219 | WEARABLE SHOOTER LOCALIZATION SYSTEM - A wearable shooter localization system including a microphone array, processor, and output device for determining information about a gunshot. The microphone array may be worn by on the upper arm of the user. A second array, which may operate cooperatively or independently from the first array, may be worn on the other arm. The microphone array is sensitive to the acoustic effects of gunfire and provides a set of electrical signals to the processing unit, which identifies the origin of the fire. The system may include orientation and/or motion detection sensors, which the processor may use to either initially compute a direction to the origin of a projectile in a frame of reference meaningful to a wearer of the system or to subsequently update that direction as the wearer moves. | 07-12-2012 |
20120207322 | MICROPHONE ARRAY WITH REAR VENTING - Microphone arrays (MAs) are described that position and vent microphones so that performance of a noise suppression system coupled to the microphone array is enhanced. The MA includes at least two physical microphones to receive acoustic signals. The physical microphones make use of a common rear vent (actual or virtual) that samples a common pressure source. The MA includes a physical directional microphone configuration and a virtual directional microphone configuration. By making the input to the rear vents of the microphones (actual or virtual) as similar as possible, the real-world filter to be modeled becomes much simpler to model using an adaptive filter. | 08-16-2012 |
20120207323 | METHOD AND APPARATUS FOR SOUND SOURCE LOCALIZATION USING MICROPHONES - A method and apparatus for sound source localization using microphones are disclosed. The method includes: receiving signals coming from a sound source through microphones covering all directions; distinguishing the received signals into those signals directly input to the microphones from the sound source (direct signals) and those signals indirectly input to the microphones (indirect signals); identifying a candidate region at which the sound source is present using locations of the microphones receiving direct signals; selecting a point in the candidate region as a candidate location; drawing one or more virtual tangent lines, contacting with the circumference of the apparatus, from the candidate location; placing locations of the microphones receiving indirect signals on the virtual tangent lines; and localizing the sound source on the basis of signals passing through the microphones receiving direct signals and through the virtual locations of the microphones receiving indirect signals. | 08-16-2012 |
20120207324 | Multiple Microphone System - A microphone system has a primary microphone for producing a primary signal, a secondary microphone for producing a secondary signal, and a selector operatively coupled with both the primary microphone and the secondary microphone. The system also has an output for delivering an output audible signal principally produced by one of the to microphones. The selector selectively permits either 1) at least a portion of the primary signal and/or 2) at least a portion of the secondary signal to be forwarded to the output as a function of the noise in the primary signal. | 08-16-2012 |
20120224714 | Host mode for an audio conference phone - A system and method for receiving sound from a teleconference host at a teleconference phone is disclosed. The method comprises identifying a person to act as the teleconference host. A location of the identified teleconference host relative to the teleconference phone is determined. A plurality of microphones on the conference phone are configured as a beamforming receiver to receive an audio signal from the location of the teleconference host. Selected microphones from the plurality of microphones are biased to receive sound from the direction of the teleconference host relative to sound received from other directions. | 09-06-2012 |
20120224715 | Noise Adaptive Beamforming for Microphone Arrays - The subject disclosure is directed towards a noise adaptive beamformer that dynamically selects between microphone array channels, based upon noise energy floor levels that are measured when no actual signal (e.g., no speech) is present. When speech (or a similar desired signal) is detected, the beamformer selects which microphone signal to use in signal processing, e.g., corresponding to the lowest noise channel. Multiple channels may be selected, with their signals combined. The beamformer transitions back to the noise measurement phase when the actual signal is no longer detected, so that the beamformer dynamically adapts as noise levels change, including on a per-microphone basis, to account for microphone hardware differences, changing noise sources, and individual microphone deterioration. | 09-06-2012 |
20120224716 | SOUND PICKUP DEVICE - A sound pickup device is provided that includes a first housing, a second housing, a first microphone, and a second microphone. The second housing is coupled to the first housing and is configured to change positions with respect to the first housing. The first microphone is mounted on the first housing and is configured to output a first audio signal based on sound picked up by the first microphone. The second microphone is mounted on the second housing and is configured to output a second audio signal based on sound picked up by the second microphone. | 09-06-2012 |
20120230511 | MICROPHONE ARRAY WITH REAR VENTING - Techniques for noise suppression systems coupled to one or more microphone arrays are described, including a housing, a first microphone, a second microphone, and a third microphone, where the third microphone functions as a common rear vent for the first and the second microphones. | 09-13-2012 |
20120230512 | Audio Zooming Process within an Audio Scene - A method comprising: obtaining a plurality of audio signals originating from a plurality of audio sources in order to create an audio scene; analyzing the audio scene in order to determine zoomable audio points within the audio scene; and providing information regarding the zoomable audio points to a client device for selecting. | 09-13-2012 |
20120237055 | METHOD FOR DUBBING MICROPHONE SIGNALS OF A SOUND RECORDING HAVING A PLURALITY OF MICROPHONES - In order to compensate tonal changes arising from a multi-path propagation of sound portions during the mixing of multi microphone audio recordings as far as possible it is suggested to form spectral values of respectively overlapping time frames of samples of each a first microphone signal ( | 09-20-2012 |
20120250881 | MICROPHONE BIASING - A plurality of microphones are coupled in series to receive a bias current. A plurality of configurable switches may be used to select which ones of the microphones receive the bias current. The current source may be adjustable and the switches may be reconfigurable to dynamically change both the number of microphones being used and the amount of bias current being generated. | 10-04-2012 |
20120263315 | SOUND SIGNAL PROCESSING DEVICE, METHOD, AND PROGRAM - There is provided a sound signal processing device, in which an observation signal analysis unit receives multi-channels of sound-signals acquired by a sound-signal input unit and estimates a sound direction and a sound segment of a target sound to be extracted and a sound source extraction unit receives the sound direction and the sound segment of the target sound and extracts a sound-signal of the target sound. By applying short-time Fourier transform to the incoming multi-channel sound-signals this device generates an observation signal in the time-frequency domain and detects the sound direction and the sound segment of the target sound. Further, based on the sound direction and the sound segment of the target sound, this device generates a reference signal corresponding to a time envelope indicating changes of the target's sound volume in the time direction, and extracts the signal of the target sound, utilizing the reference signal. | 10-18-2012 |
20120275619 | AUDIO SIGNAL PROCESSING APPARATUS, AUDIO SIGNAL PROCESSING METHOD AND IMAGING APPARATUS - An audio signal processing apparatus generates an audio signal having an omni-directivity in the whole circumferential direction, generates an audio signal having a directivity in the right-left direction, generates an audio signal having a directivity in the front-back direction, adds the audio signal resulting from the multiplication of the audio signal having a directivity in the whole circumferential direction by a predetermined coefficient, the audio signal resulting from the multiplication of the audio signal having a directivity in the right-left direction by a predetermined coefficient, and the audio signal resulting from the multiplication of the audio signal having a directivity in the front-back direction by a predetermined coefficient, and generates a unidirectional audio signal. | 11-01-2012 |
20120275620 | MICROPHONE ARRAY APPARATUS AND STORAGE MEDIUM STORING SOUND SIGNAL PROCESSING PROGRAM - A microphone array apparatus includes: an acquisition unit configured to acquire samples from a sound signal inputted from each of a plurality of microphones, at predetermined time intervals; an operation unit configured to calculate a value based on volumes of the sound signal possessed by a plurality of the samples for each of the sound signals inputted from the plurality of microphones; a correlation coefficient calculator configured to calculate a coefficient of correlation between the sound signals, on the basis of the values calculated for the respective sound signals; and a gain calculator configured to calculate reduction gain for the sound signals inputted from the plurality of microphones, on the basis of the coefficient of correlation. | 11-01-2012 |
20120275621 | Surface-Mounted Microphone Arrays on Flexible Printed Circuit Boards - A microphone array, having a three-dimensional (3D) shape, has a plurality of microphone devices mounted onto (at least one) flexible printed circuit board (PCB), which is bent to achieve the 3D dimensional shape. Output signals from the microphone devices can be combined (e.g., by weighted or unweighted summation or differencing) to form sub-element output signals and/or element output signals, and ultimately a single array output signal for the microphone array. The PCB may be uniformly flexible or may have rigid sections interconnected by flexible portions. Possible 3D shapes include (without limitation) cylinders, spirals, serpentines, and polyhedrons, each formed from a single flexible PCB. Alternatively, the microphone array may be an assembly of multiple, interconnecting sub-arrays, each having two or more rigid portions separated by one or more flexible portions, where each sub-array has at least one cut-out portion for receiving a rigid portion of another sub-array. | 11-01-2012 |
20120281853 | SYSTEM AND METHOD FOR ENHANCING SPEECH INTELLIGIBILITY USING COMPANION MICROPHONES WITH POSITION SENSORS - Systems and methods for enhancing speech intelligibility using a companion microphone system can include microphones, a position sensor and a microcontroller. In certain embodiments, the position sensor is configured to generate position data corresponding to a position of the companion microphone system. In various embodiments, the microphones and the position sensor include a fixed relationship in three-dimensional space. In certain embodiments, the microcontroller is configured to receive the position data from the position sensor and select one or more of the microphones to receive an audio input based on the received position data. | 11-08-2012 |
20120281854 | SOUND EMISSION AND COLLECTION DEVICE - A sound emission and collection device has a plurality of speakers ( | 11-08-2012 |
20120288113 | MICROPHONE - A microphone includes a plurality of microphone units; in which the microphone units include a first group of microphone units and a second group of microphone units, the first group of microphone units and the second group of microphone units are disposed alternately, the first group of microphone units are connected in series such that outputs from the first group of microphone units are added and outputted as an added output, the second group of microphone units are connected in series such that outputs from the second group of microphone units are added and outputted as another added output, and the added output of one of the first group of microphone units and the second group of microphone units is output from a hot terminal as a balanced output and the other added output is output from a cold terminal as a balanced output. | 11-15-2012 |
20120288114 | AUDIO CAMERA USING MICROPHONE ARRAYS FOR REAL TIME CAPTURE OF AUDIO IMAGES AND METHOD FOR JOINTLY PROCESSING THE AUDIO IMAGES WITH VIDEO IMAGES - A method comprises providing at least one processing unit comprising a decomposing section and a playback section; receiving, at the decomposing section, audio data generated via an array of microphones, the audio data representing an acoustic scene; decomposing the audio data into a plurality of signals representing components of the acoustic scene arriving from a plurality of directions, using the decomposing section; and rendering the audio components for a listener based on the plurality of directions of the audio components, using the playback section. | 11-15-2012 |
20120294456 | SIGNAL SOURCE LOCALIZATION USING COMPRESSIVE MEASUREMENTS - In one aspect, a method for performing signal source localization is provided. The method comprises the steps of obtaining compressive measurements of an acoustic signal or other type of signal from respective ones of a plurality of sensors, processing the compressive measurements to determine time delays between arrivals of the signal at different ones of the sensors, and determining a location of a source of the signal based on differences between the time delays. The method may be implemented in a processing device that is configured to communicate with the plurality of sensors. In an illustrative embodiment, the compressive measurements are obtained from respective ones of only a designated subset of the sensors, and a non-compressive measurement is obtained from at least a given one of the sensors not in the designated subset, with the time delays between the arrivals of the signal at different ones of the sensors being determined based on the compressive measurements and the non-compressive measurement. | 11-22-2012 |
20120308037 | MICROELECTROMECHANICAL MICROPHONE CHIP HAVING STEREOSCOPIC DIAPHRAGM STRUCTURE AND FABRICATION METHOD THEREOF - A microelectromechanical microphone chip having a stereoscopic diaphragm structure includes a base, having a chamber; a diaphragm, disposed on the chamber and having steps with height differences; and a back plate, disposed on the diaphragm, forming a space with the diaphragm in between, and having a plurality of sound-holes communicating with the space. | 12-06-2012 |
20120308038 | Sound Source Localization Apparatus and Method - Sound source localization apparatuses and methods are described. A frame amplitude difference vector is calculated based on short time frame data acquired through an array of microphones. The frame amplitude difference vector reflects differences between amplitudes captured by microphones of the array during recording the short time frame data. Similarity between the frame amplitude difference vector and each of a plurality of reference frame amplitude difference vectors is evaluated. Each of the plurality of reference frame amplitude difference vectors reflects differences between amplitudes captured by microphones of the array during recording sound from one of a plurality of candidate locations. A desired location of sound source is estimated based at least on the candidate locations and associated similarity. The sound source localization can be performed based at least on amplitude difference. | 12-06-2012 |
20120308039 | SOUND SOURCE SEPARATION SYSTEM, SOUND SOURCE SEPARATION METHOD, AND ACOUSTIC SIGNAL ACQUISITION DEVICE - A sound source separation system separates a target sound and a disturbance sound coming from an arbitrary direction other than the direction the target sound comes from. The system includes different-directional-signal-group generators and a sensitive region formation unit. The generators each generate two or more combinations of spectra of signals each of which has a different directivity, using received sound signals of microphones. The sensitive region formation unit determines, for each frequency band, whether or not a relationship between powers of the spectra in each combination simultaneously satisfies conditions each defined for each combination, using two or more combinations of the spectra of the signals generated by the respective different-directional-signal-group generators, and performs multidimensional band selection of assigning power of a spectrum selected beforehand to a spectrum of the target sound to be separated, for a frequency band where the conditions are simultaneously satisfied. | 12-06-2012 |
20120308040 | MICROPHONE ARRAY CALIBRATION METHOD AND APPARATUS - An apparatus for providing real-time calibration for two or more microphones. A calibrator for receiving a left microphone signal and a right microphone signal and generating phase difference data. A phase and amplitude correction system for receiving one of the left microphone signal or the right microphone signal the phase difference data and generating calibration data for a beamformer. The beamformer receiving the calibration data, the left microphone signal and the right microphone signal and generating a monaural beamformed signal. | 12-06-2012 |
20120314885 | SIGNAL PROCESSING USING SPATIAL FILTER - A device and method processing microphone signals from at least two microphones is presented. A first beamformer processes the signals from the microphones and provides a first beamformed signal. A power estimator processes the signals from the microphones and the first beamformed signal from the first beamformer in order to generate, in frequency bands, a first statistical estimate of the energy of a first part of an incident sound field. A gain controller processes said first statistical estimate in order to generate in frequency bands a first gain signal, and an audio processor for processing an input to the signal processing device in dependence of said generated first gain signal. The invention provides a new and improved noise reduction device and noise reduction method for use in the signal processing in devices processing acoustic signals, e.g. microphone devices. | 12-13-2012 |
20120321100 | Wide Dynamic Range Microphone - A microphone system has an output and at least a first transducer with a first dynamic range, a second transducer with a second dynamic range different than the first dynamic range, and coupling system to selectively couple the output of one of the first transducer or the second transducer to the system output, depending on the magnitude of the input sound signal, to produce a system with a dynamic range greater than the dynamic range of either individual transducer. A method of operating a microphone system includes detecting whether a transducer output crosses a threshold, and if so then selectively coupling another transducer's output to the system output. Some embodiments combine the outputs of more than one transducer in a weighted sum during transition from one transducer output to another, as a function of time or as a function of the amplitude of the incident audio signal. | 12-20-2012 |
20130010980 | VEHICLE DIRECTION IDENTIFICATION DEVICE, VEHICLE DIRECTION IDENTIFICATION METHOD AND PROGRAM THEREFOR - A vehicle direction identification device includes: a frequency analysis unit which analyzes phase of the surrounding sound in each analysis section specified by predetermined frequency regions and time intervals; a sound source direction identification unit which identifies a sound source direction indicating a direction of a sound included in the vehicle sound for each analysis section; a reflection information storage unit which stores (i) state information relating to rates of occurrence each of which are a count of the analysis sections of a corresponding one of the sound source directions, and (ii) reflection patterns each of which includes an estimated vehicle direction which is a vehicle direction associated with a set of the state information; and a vehicle direction identification unit which identifies a vehicle direction by checking the rates of occurrence obtained from an identification result by the sound source direction identification unit against one of the reflection patterns. | 01-10-2013 |
20130016852 | SOUND SOURCE LOCALIZATION USING PHASE SPECTRUMAANM Regunathan; ShankarAACI RedmondAAST WAAACO USAAGP Regunathan; Shankar Redmond WA USAANM Koishida; KazuhitoAACI RedmondAAST WAAACO USAAGP Koishida; Kazuhito Redmond WA USAANM Kikkeri; Harshavardhana NarayanaAACI BellevueAAST WAAACO USAAGP Kikkeri; Harshavardhana Narayana Bellevue WA US - An array of microphones placed on a mobile robot provides multiple channels of audio signals. A received set of audio signals is called an audio segment, which is divided into multiple frames. A phase analysis is performed on a frame of the signals from each pair of microphones. If both microphones are in an active state during the frame, a candidate angle is generated for each such pair of microphones. The result is a list of candidate angles for the frame. This list is processed to select a final candidate angle for the frame. The list of candidate angles is tracked over time to assist in the process of selecting the final candidate angle for an audio segment. | 01-17-2013 |
20130022216 | SYSTEMS AND METHODS FOR PROCESSING AUDIO SIGNALS CAPTURED USING MICROPHONES OF MULTIPLE DEVICES - Systems, methods and apparatus for capturing at least one audio signal using a plurality of microphones that generate a plurality of representations of the at least one audio signal. In some embodiments, the plurality of microphones are disposed in a multiple-microphone setting so that the at least one audio signal is captured by at least two of the plurality of microphones. In some embodiments, at least one of the plurality of microphones is a microphone of a mobile device. The plurality of representations of the at least one audio signal may be processed to obtain a processed representation of the at least one audio signal. | 01-24-2013 |
20130022217 | SOUND ZOOM METHOD, MEDIUM, AND APPARATUS - A sound zoom method, medium, and apparatus generating a signal in which a target sound is removed from sound signals input to a microphone array by adjusting a null width that restricts a directivity sensitivity of the microphone array, and extracting a signal corresponding to the target sound from the sound signals by using the generated signal. Thus, a sound located at a predetermined position away from the microphone array can be selectively obtained so that a target sound is efficiently obtained. | 01-24-2013 |
20130028439 | INPUT DEVICE, SIGNAL PROCESSING METHOD, PROGRAM, AND RECORDING MEDIUM - There is provided an input device including at least two microphones placed at different positions on a chassis to face different directions on one of space axes, a low-frequency bandwidth extracting part for extracting a low-frequency bandwidth signal from a signal input from the microphones, a phase difference calculating part for calculating a phase difference using the low-frequency bandwidth signal extracted by the low-frequency bandwidth extracting part; and a control signal generating part for generating a control signal based on the phase difference calculated by the phase difference calculating part. | 01-31-2013 |
20130034241 | METHODS AND APPARATUSES FOR MULTIPLE CONFIGURATIONS OF BEAMFORMING MICROPHONE ARRAYS - Embodiments include methods and apparatuses for sensing acoustic waves for a conferencing application. A conferencing apparatus includes a plurality of directional microphones oriented to cover a corresponding plurality of direction vectors and disposed in a housing. An orientation sensor is configured to generate an orientation signal indicative of an orientation of the housing. A processor is operably coupled to the plurality of directional microphones and the orientation sensor. The processor is configured to automatically adjust a signal processing characteristic of one or more directional microphones of the plurality of directional microphones responsive to the orientation signal. | 02-07-2013 |
20130044893 | System and method for muting audio associated with a source - In one embodiment, a method includes receiving audio at a plurality of microphones, identifying a sound source to be muted, processing the audio to remove sound received from the sound source at each of the microphones, and transmitting the processed audio. An apparatus is also disclosed. | 02-21-2013 |
20130044894 | SYSTEM AND METHOD FOR EFFICIENT SOUND PRODUCTION USING DIRECTIONAL ENHANCEMENT - A system and method for generating virtual microphone signals having a particular number and configuration for channel playback from an intermediate set of signals that were recorded in an initial format that is different from the channel playback format. In one embodiment, an initial set of intermediate are Bark-banded such that each intermediate signal may lead to a corresponding power spectral density (PSD) signal representative of the initial intermediate signal. Further, one may generate cross-correlations signals for each pair of intermediate signals. Next, from the PSDs and cross correlations, one may more efficiently calculate corresponding channel signals to be used for playback on respective channel speakers. Thus, the PSDs of each channel signal may be generated at chosen angles (as well as other design factors). Further, each channel signal may also be further modified with a corresponding cancellation signal that further enhances the resultant signal in each channel. | 02-21-2013 |
20130051577 | ARRAY MICROPHONE APPARATUS FOR GENERATING A BEAM FORMING SIGNAL AND BEAM FORMING METHOD THEREOF - Embodiments described in the present disclosure relate to an array microphone apparatus for generating a beam forming signal. The apparatus includes first, second, and third omni-directional microphones, each converting an audible signal into a corresponding electrical signal. The second omni-directional microphone is disposed between the other two omni-directional microphones. The apparatus includes a first directional microphone forming device to jointly output a first directional microphone signal with a first bi-directional pattern, and a magnitude and phase response handler device to output a second directional microphone signal with an omni-directional pattern shifted by a prefixed value with respect to first directional microphone signal. The apparatus further includes a combining device receiving the first and second directional microphone signals and outputting a combined directional microphone signal with a combined beam pattern correlated to the first bi-directional and second omni-directional patterns, the combined directional microphone signal being in a broadside configuration. | 02-28-2013 |
20130064391 | Acoustic Beam Forming Array Using Feedback-Controlled Microphones for Tuning and Self-Matching of Frequency Response - A feedback-controlled microphone includes a microphone body and a membrane operatively connected to the body. The membrane is configured to be initially deflected by acoustic pressure such that the initial deflection is characterized by a frequency response. The microphone also includes a sensor configured to detect the frequency response of the initial deflection and generate an output voltage indicative thereof. The microphone additionally includes a compensator in electric communication with the sensor and configured to establish a regulated voltage in response to the output voltage. Furthermore, the microphone includes an actuator in electric communication with the compensator, wherein the actuator is configured to secondarily deflect the membrane in opposition to the initial deflection such that the frequency response is adjusted. An acoustic beam forming microphone array including a plurality of the above feedback-controlled microphones is also disclosed. | 03-14-2013 |
20130070938 | NOISE CANCELLING DEVICE - A noise cancelling device includes an extracting unit configured to extract a first noise from a signal, the signal being based on an input audio signal, a storing unit configured to store noise characteristic information on a second noise, the second noise remaining after subtracting the extracted first noise from the signal based on the audio signal. And the device further includes a cancelling unit configured to perform cancelling processing for cancelling a noise on the input audio signal based on the first noise and the noise characteristic information on the second noise. | 03-21-2013 |
20130083942 | Processing Signals - Beamformer coefficients may include a plurality of sets of theoretical statistical data for theoretical signals. Each theoretical signal may have its own particular attributes. The statistical data may be used in computing beamformer coefficients for application by a beamformer to signals received at a device. Signals are received at an input of the device. A respective plurality of weights is determined, for the theoretical statistical data sets, based on an analysis of the extent to which the signals have the particular attributes of the theoretical signals. The theoretical statistical data sets are retrieved, and a statistical data set is calculated for the signals by performing a weighted sum of the theoretical statistical data sets using the determined respective plurality of weights. Beamformer coefficients are computed based on the calculated statistical data set for the signals, which are used by a beamformer to the signals for generating a beamformer output. | 04-04-2013 |
20130083943 | Processing Signals - Method, device and computer program product for processing signals at the device. Signals are received, over a range of angles, at a plurality of sensors of the device, the received signals including an interfering signal received from an interfering source location. An interference delay pattern between receipt of signals at the sensors corresponding to receipt of a signal from the interfering source location is determined. A plurality of regularization signals having a delay pattern matching the determined interference delay pattern are generated. The generated regularization signals are used to determine beamformer coefficients to be applied by a beamformer, and the beamformer applies the determined beamformer coefficients to the signals received by the plurality of sensors, thereby generating a beamformer output. | 04-04-2013 |
20130083944 | APPARATUS - An apparatus comprising at least one processor and at least one memory including computer program code the at least one memory and the computer program code configured to, with the at least one processor, cause the apparatus at least to perform determining a change of position of the apparatus, and processing at least one audio signal dependent on the change in position. | 04-04-2013 |
20130089218 | AUDIO EQUIPMENT AND OSCILLATION UNIT - A mobile phone ( | 04-11-2013 |
20130094664 | METHOD AND DEVICE FOR PHASE-SENSITIVE PROCESSING OF SOUND SIGNALS - A method and device for phase-sensitive processing of sound signals of at least one sound source may include arranging two microphones at a distance d from each other, capturing sound signals with both microphones, generating associated microphone signals, and processing the sound signals of the microphones. During a calibration mode, a calibration-position-specific, frequency-dependent phase difference vector φ0(f) between the associated calibration microphone signals may be calculated from their frequency spectra for the calibration position. Then, during an operating mode, a signal spectrum S of a signal to be output is calculated by multiplication of at least one of the two frequency spectra of the current microphone signals with a spectral filter function F. | 04-18-2013 |
20130101136 | WEARABLE DIRECTIONAL MICROPHONE ARRAY APPARATUS AND SYSTEM - 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. | 04-25-2013 |
20130108074 | MICROPHONE | 05-02-2013 |
20130121504 | MICROPHONE ARRAY WITH DAISY-CHAIN SUMMATION - Microphone stages in a microphone array may be coupled together in a daisy chain. Each stage may include a microphone, an analog to digital converter, a decimation unit, a receiver, an adder, and a transmitter. The converter may convert analog audio microphone signals into digital codes that may be decimated. The adder may add decimated digital codes in each stage to a cumulative sum of decimated digital codes from prior stages. This new sum may be transmitted to the next microphone stage, where the adder may add the decimated digital codes from that stage to the cumulative sum. A serial interface may be used to connect the transmitters and receivers of each of the stages. The serial interface may be used to transmit the cumulative sum of decimated digital codes between the stages. The serial interface may also be used to transmit configuration data between the stages. | 05-16-2013 |
20130121505 | MICROPHONE ARRAY CONFIGURATION AND METHOD FOR OPERATING THE SAME - An apparatus comprises a plurality of microphone units including at least a first microphone unit and a second microphone unit, each of the first and second microphone units comprising a microphone, an analog-to-digital converter, and a local memory. The microphone is configured to capture an analog audio signal. The analog-to-digital converter is configured to convert the analog audio signal created by the microphone into a digital audio signal. The local memory is configured to store the digital audio signal. The apparatus further comprises, a controller unit comprising a processor configured to process the digital audio signals. The first microphone unit and the second microphone unit are operatively connected to the controller unit in a series configuration, the second microphone unit being configured to output the digital audio signal to the first microphone unit, and the first microphone unit being configured to output the digital audio signal to the controller unit. | 05-16-2013 |
20130129113 | SOUND SOURCE SIGNAL FILTERING APPARATUS BASED ON CALCULATED DISTANCE BETWEEN MICROPHONE AND SOUND SOURCE - Provided is a sound source signal filtering method and apparatus. The sound source signal filtering method includes: generating two or more microphone output signals by combining sound source signals input through a plurality of microphones; calculating distances between the microphones and a sound source from which the sound source signals are emitted by using distance relationships according to frequencies of the sound source signals extracted from the generated microphone output signals; and filtering the sound source signals to obtain one or more sound source signals corresponding to a predetermined distance by using the calculated distances. Accordingly, it is possible to obtain only sound source signals emitted from a sound source at a particular distance from the microphone array among a plurality of sound source signals input through the microphone array. | 05-23-2013 |
20130136273 | 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. | 05-30-2013 |
20130136274 | Processing Signals - Method, device and computer program product for processing signals. Signals are received at a plurality of sensors of the device. The initiation of a signal state in which signals of a particular type are received at the plurality of sensors is determined. Responsive to the determining of the initiation of the signal state, data indicating beamformer coefficients to be applied by a beamformer of the device is retrieved from data storage means, wherein the indicated beamformer coefficients are determined so as to be suitable for application to signals received at the sensors in the signal state. The beamformer applies the indicated beamformer coefficients to the signals received at the sensors in the signal state, thereby generating a beamformer output. | 05-30-2013 |
20130142355 | NEAR-FIELD NULL AND BEAMFORMING - Devices and methods are disclosed that allow for selective acoustic near-field nulls for microphone arrays. One embodiment may take the form of an electronic device including a speaker and a microphone array. The microphone array may include a first microphone positioned a first distance from the speaker and a second microphone positioned a second distance from the speaker. The first and second microphones are configured to receive an acoustic signal. The microphone array further includes a complex vector filter coupled to the second microphone. The complex vector filter is applied to an output signal of the second microphone to generate an acoustic sensitivity pattern for the array that provides an acoustic null at the location of the speaker. | 06-06-2013 |
20130142356 | NEAR-FIELD NULL AND BEAMFORMING - Devices and methods are disclosed that allow for selective acoustic near-field nulls for microphone arrays. One embodiment may take the form of an electronic device including a speaker and a microphone array. The microphone array may include a first microphone positioned a first distance from the speaker and a second microphone positioned a second distance from the speaker. The first and second microphones are configured to receive an acoustic signal. The microphone array further includes a complex vector filter coupled to the second microphone. The complex vector filter is applied to an output signal of the second microphone to generate an acoustic sensitivity pattern for the array that provides an acoustic null at the location of the speaker. | 06-06-2013 |
20130142357 | METHOD FOR VISUALIZING SOUND SOURCE ENERGY DISTRIBUTION IN ECHOIC ENVIRONMENT - A method for visualizing sound source energy distribution in an echoic environment comprises steps: arranging in an echoic environment a plurality of arrayed sound pickup units, wherein each sound pickup unit includes at least two microphones separated by a directive distance enabling the sound pickup unit to have a primary pickup direction; disposing the sound pickup units with the primary pickup directions thereof pointing toward a sound source in the echoic environment, and measuring the sound source by the sound pickup units to obtain a sound source-related parameter; substituting the directive distance and the parameter into an algorithm to make the parameter have directivity; and then substituting the parameter into an ESM algorithm to establish a sound source energy distribution profile. Thereby, the method can measure a sound source in a specified direction in an echoic environment and establish a visualized sound source energy distribution profile. | 06-06-2013 |
20130142358 | Variable Directivity MEMS Microphone - The directivity pattern of a MEMS microphone is adjusted. An indication of a position of a MEMS microphone is received and the microphone includes at least one diaphragm. An adjustment to the position of the MEMS microphone is determined. Based upon the adjustment, a position of the at least one diaphragm is adjusted. The adjusting is effective to alter a directivity pattern of the microphone. | 06-06-2013 |
20130156220 | SELECTIVE SPATIAL AUDIO COMMUNICATION - Audio data associated with a plurality of originating sources is obtained, the audio data directed to a participant entity. An originating entity associated with one of the originating sources is determined. A listener focus indication is obtained from the participant entity indicating a listener focus on the originating entity. A spatial positional relationship is determined between the participant and originating entities. A filtering operation is initiated to enhance a portion of the audio data associated with the originating entity, the portion enhanced relative to another portion of the audio data that is associated with the originating sources other than the first one. A spatialization of a stream of the first portion that is based on a participant positional listening perspective is initiated, based on the spatial positional relationship. Transmission of a spatial stream of audio data is initiated to the participant entity, based on the filtering operation and spatialization. | 06-20-2013 |
20130156221 | SIGNAL PROCESSING APPARATUS AND SIGNAL PROCESSING METHOD - A signal processing apparatus includes an adder that acquires a plurality of input signals from a plurality of microphones and calculates an added value obtained by adding the input signals together, a subtracter that acquires a plurality of input signals from the plurality of microphones and calculates a subtracted value obtained by subtracting one input signal from the other input signal, and a determination unit that determines whether noise is included in the input signals based on the added value and the subtracted value. | 06-20-2013 |
20130170666 | ADAPTIVE SELF-CALIBRATION OF SMALL MICROPHONE ARRAY BY SOUNDFIELD APPROXIMATION AND FREQUENCY DOMAIN MAGNITUDE EQUALIZATION - Methods and apparatus for self-calibration of small-microphone arrays are described. In one embodiment, self-calibration is based upon a mathematical approximation for which a detected response by one microphone should approximately equal a combined response from plural microphones in the array. In a second embodiment, self-calibration is based upon matching gains in each of a plurality of Bark frequency bands, and applying the matched gains to frequency domain microphone signals such that the magnitude response of all the microphones in the array approximates an average magnitude response for the array. The methods and apparatus may be implemented in hearing aids or small audio devices and used to mitigate adverse aging and mechanical effects on acoustic performance of small-microphone arrays in these systems. | 07-04-2013 |
20130177168 | Apparatus - An apparatus comprising at least one processor and at least one memory including computer program code the at least one memory and the computer program code configured to, with the at least one processor, cause the apparatus at least to perform determining a change in position and/or orientation of an apparatus, and processing at least two audio signals dependent on the change in position and/or orientation to generate at least one output signal wherein the processing of the two audio signals dependent on the change in position and/or orientation produces the output signal comprising a representation of acoustic energy from a first direction. | 07-11-2013 |
20130195285 | ZONE BASED PRESENCE DETERMINATION VIA VOICEPRINT LOCATION AWARENESS - A speech from a speaker proximate to one or more microphones within an environment can be received. The microphones can be a directional microphone or an omni-directional microphone. The speech can be processed to produce an utterance to determine the identity of the speaker. The identity of the speaker can be associated with a voiceprint. The identity can be associated with a user's credentials of a computing system. The credentials can uniquely identify the user within the computing system. The utterance can be analyzed to establish a zone in which the speaker is present. The zone can be a bounded region within the environment. The zone can be mapped within the environment to determine a location of the speaker. The location can be a relative or an absolute location. | 08-01-2013 |
20130223644 | Systems and Methods for Reducing Unwanted Sounds in Signals Received From an Arrangement of Microphones - A system and method for reducing unwanted sounds in signals received from an arrangement of microphones including: sensing sound sources distributed around a specified target direction by way of an arrangement of microphones to produce left and right microphone output signals; determining the power of the left and right microphone signals; determining the minimum of the two microphone power measures; and, attenuating the signals based on a comparison of the left and right microphone power measures with the minimum power measure. | 08-29-2013 |
20130230186 | Electronic Device And Direction Switching Method Of The Electronic Device - An electronic device and a direction switching method are described. The electronic device includes a housing having a first end and a second end, an audio input unit and an audio output unit. The electronic device has at least a first working direction and a second working direction, the direction from said first end to said second end is a reference direction when the electronic device is in the first working direction; and the direction from the second end to the first end is the reference direction when the electronic device is in the second working direction, wherein the audio input unit in a working state is located at the second end when the electronic device is in the first working direction; and the audio input unit in the working state is located at the first end when the electronic device is in the second working direction. | 09-05-2013 |
20130230187 | APPARATUS AND METHOD FOR DERIVING A DIRECTIONAL INFORMATION AND COMPUTER PROGRAM PRODUCT - An apparatus for deriving a directional information from a plurality of microphone signals or from a plurality of components of a microphone signal, wherein different effective microphone look directions are associated with the microphone signals or components, has a combiner configured to obtain a magnitude value from a microphone signal or a component of the microphone signal. The combiner is further configured to combine direction information items describing the effective microphone look directions, such that a direction information item describing a given effective microphone look direction is weighted in dependence on the magnitude value of the microphone signal, or of the component of the microphone signal, associated with the given effective microphone look direction, to derive the directional information. | 09-05-2013 |
20130259262 | SOUND RECORDING DEVICE - A sound recording device of the present disclosure is can be connected with an external sound pickup device. The sound recording device includes a connector having a plurality of terminals to which the external sound pickup device can be connected, and a determiner that determines a type of the external sound pickup device when the external sound pickup device is connected to the connector, based on a correlation between signals of specific terminals out of the plurality of terminals. | 10-03-2013 |
20130272538 | SYSTEMS, METHODS, AND APPARATUS FOR INDICATING DIRECTION OF ARRIVAL - Systems, methods, and apparatus for projecting an estimated direction of arrival of sound onto a plane that does not include the estimated direction are described. | 10-17-2013 |
20130272539 | SYSTEMS, METHODS, AND APPARATUS FOR SPATIALLY DIRECTIVE FILTERING - Systems, methods, and apparatus are described for applying, based on angles of arrival of source components relative to the axes of different microphone pairs, a spatially directive filter to a multichannel audio signal to produce an output signal. | 10-17-2013 |
20130287223 | UNIDIRECTIONAL MICROPHONE - The present invention provides a unidirectional microphone by adding an output from an omnidirectional condenser microphone unit and an output from a bi-directional ribbon microphone unit together. A condenser microphone unit | 10-31-2013 |
20130287224 | NOISE SUPPRESSION BASED ON CORRELATION OF SOUND IN A MICROPHONE ARRAY - A microphone array includes a left microphone, a right microphone and a processor to receive a right microphone signal from the right microphone and a left microphone signal from the left microphone. The processor determines a timing difference between the left microphone signal and the right microphone signal. The processor determines whether the timing difference is within a time threshold. The processor time shifts one of the left microphone signal and the right microphone signal based on the timing difference. The processor also sums the shifted microphone signal and the other microphone signal to form an output signal. | 10-31-2013 |
20130287225 | SOUND ENHANCEMENT METHOD, DEVICE, PROGRAM AND RECORDING MEDIUM - A sound enhancement technique that uses transfer functions a | 10-31-2013 |
20130308790 | METHODS AND SYSTEMS FOR DOPPLER RECOGNITION AIDED METHOD (DREAM) FOR SOURCE LOCALIZATION AND SEPARATION - Systems and methods are provided for source localization and separation by sampling a large scale microphone array asynchronously to simulate a smaller size but moving microphone array. Signals that arrive from different angles at the array are shifted differently in their frequency content. The sources are separated by evaluating correlated and even equal frequency content. Compressive sampling enables the utilization of extremely large scale microphone arrays by reducing the computational effort orders of magnitude in comparison to standard synchronous sampling approaches. Processor based systems to perform the source separation methods are also provided. | 11-21-2013 |
20130329907 | MINITURE ELECTRONIC SHOTGUN MICROPHONE - A miniature electronic shotgun microphone, which is used to receive a sound source from a specified direction, comprises a pick-up member, an A/D (Analog/Digital) conversion unit, and a digital signal processor. The pick-up member includes a first pick-up unit, a second pick-up unit separated from the first pick-up unit by a first distance, and a third pick-up unit separated from the second pick-up unit by a second distance; the first distance is greater than the second distance. The first pick-up unit, the second pick-up unit and the third pick-up unit respectively receive the sound source and output an analog signal. The A/D conversion unit and the digital signal processor process the analog signals, and convert them into a directional digital acoustic signal. Thus, the directional digital acoustic signal has a maximum pick-up frequency. Thereby is decreased grating lobes and spatial aliasing. | 12-12-2013 |
20130329908 | ADJUSTING AUDIO BEAMFORMING SETTINGS BASED ON SYSTEM STATE - Audio beamforming is a technique in which sounds received from two or more microphones are combined to isolate a sound from background noise. A variety of audio beamforming spatial patterns exist. The patterns can be fixed or adapted over time, and can even vary by frequency. The different patterns can achieve varying levels of success for different types of sounds. To improve the performance of audio beamforming, a system can select a mode beam pattern based on a detected running application and/or device settings. The system can use the mode beam pattern to configure an audio beamforming algorithm. The configured audio beamforming algorithm can be used to generate processed the audio data from multiple audio signals. The system can then send processed audio data to the running application. | 12-12-2013 |
20130343571 | REAL-TIME MICROPHONE ARRAY WITH ROBUST BEAMFORMER AND POSTFILTER FOR SPEECH ENHANCEMENT AND METHOD OF OPERATION THEREOF - A microphone array processing system and method carried out in the system. In one embodiment, the system includes: (1) a beamformer configured to perform adaptive beamforming on gain-compensated signals received from a plurality of microphones, the adaptive beamforming including dynamic range compression and diagonal loading of a sample correlation matrix based on order statistics and (2) a postfilter configured to receive an output of the beamformer and reduce noise components remaining from the beamforming. | 12-26-2013 |
20130343572 | MICROPHONE MOUNTING STRUCTURE OF MOBILE TERMINAL AND USING METHOD THEREOF - Disclosed are a microphone mounting structure of a mobile terminal, capable of capturing a sound generated from a subject in an optimum manner while capturing an image and reproducing the captured image, and a using method thereof. In the present invention, a first microphone and a second microphone are arranged on one side surface of a terminal body, in a spaced manner from each other on different axes, so that they can be used in a sound capturing mode. Three or more microphones are arranged on a plurality of surfaces of the terminal body, at various intervals, by interworking with various situation changes. Then, the number of microphones to be used for capturing a sound, and a microphone combination are selected according to a user's behavior scenario. Under such configuration, an optimum audio zooming function can be implemented. | 12-26-2013 |
20140016798 | MEMS DEVICE - A method of fabricating a micro-electrical-mechanical system (MEMS) apparatus on a substrate comprises the steps of processing the substrate so as to fabricate an electronic circuit; depositing a first electrode that is operably coupled with the electronic circuit; depositing a membrane so that it is mechanically coupled to the first electrode; applying a sacrificial layer; depositing a structural layer and a second electrode that is operably coupled with the electronic circuit so that the sacrificial layer is disposed between the membrane and the structural layer so as to form a preliminary structure; singulating the substrate; and removing the sacrificial layer so as to form a MEMS structure, in which the step of singulating the substrate is carried out before the step of removing the sacrificial layer. | 01-16-2014 |
20140029761 | Method and Apparatus for Microphone Beamforming - In accordance with an example embodiment of the present invention, an apparatus is disclosed. The apparatus includes a camera system and an optimization system. The optimization system is configured to communicate with the camera system. At least one microphone is connected to the optimization system. The optimization system is configured to adjust a beamform of the at least one microphone based, at least in part, on camera focus information of the camera system. | 01-30-2014 |
20140044279 | MULTI-MICROPHONE AUDIO SOURCE SEPARATION BASED ON COMBINED STATISTICAL ANGLE DISTRIBUTIONS - Systems, methods, and computer media for separating audio sources in a multi-microphone system are provided. A plurality of audio sample groups can be received. Each audio sample group comprises at least two samples of audio information captured by different microphones during a sample group time interval. For each audio sample group, an estimated angle between an audio source and the multi-microphone system can be estimated based on a phase difference of the samples in the group. The estimated angle can be modeled as a combined statistical distribution that is a mixture of a target audio signal statistical distribution and a noise component statistical distribution. The combined statistical distribution can be analyzed to provide an accurate characterization of each sample group as either target audio signal or noise. The target audio signal can then be resynthesized from samples identified as part of the target audio signal. | 02-13-2014 |
20140050332 | METHOD AND SYSTEM FOR OBTAINING AN AUDIO SIGNAL - A method and system for obtaining an audio signal. In one embodiment, the method comprises receiving a first sound signal at a first microphone arranged at a first height vertically above a substantially flat surface; receiving a second sound signal at a second microphone arranged at a second height vertically above the substantially flat surface; processing a signal provided by the first microphone using a low pass filter; processing a signal provided by the second microphone using a high pass filter; adding the signals processed by the low pass filter and the high pass filter to form a sum signal; and outputting the sum signal as an audio signal. | 02-20-2014 |
20140050333 | Data Processing Method And An Electronic Apparatus - A data processing method and an electronic apparatus are described. The hand-held electronic apparatus has a microphone array that includes at least two kinds of usage modes. The microphone array is in a first usage mode of the at least two kinds of usage modes when the hand-held electronic apparatus is in a first usage status. The method includes receiving an adjustment signal to adjust the hand-held electronic apparatus from the first usage status to a second usage status, wherein, the second usage status is different from the first usage status; obtaining a control command when the hand-held electronic apparatus is adjusted to the second usage status; adjusting the microphone array from the first usage mode to the second usage mode according to, wherein, the second usage mode is different from the first usage mode and corresponds to the second usage status. | 02-20-2014 |
20140056439 | ELECTRONIC DEVICE AND METHOD FOR SELECTING MICROPHONE BY DETECTING VOICE SIGNAL STRENGTH - An electronic device and method thereof for selecting a mic by detecting voice signal strengths at respective microphones. The electronic device has first and second mics and is communicatively connected to a headset having a third mic. Voice signal strengths received at the respective first, second, and third mics are detected. A determination is made as to which one of the first, second, and third mics detects the greatest voice signal strength. All of the mics are disabled for communication except for the mic detecting the greatest voice signal strength. | 02-27-2014 |
20140072142 | SOUND DIRECTION ESTIMATION DEVICE, SOUND PROCESSING SYSTEM, SOUND DIRECTION ESTIMATION METHOD, AND SOUND DIRECTION ESTIMATION PROGRAM - A sound direction estimation device includes a first correlation matrix calculation unit configured to calculate a correlation matrix of a plurality of channels of input sound signals, a second correlation matrix calculation unit configured to calculate a correlation matrix of noise signals based on the plurality of channels of sound signals, and a sound source localization unit configured to calculate a spatial spectrum based on the correlation matrix calculated by the first correlation matrix calculation unit and the correlation matrix calculated by the second correlation matrix calculation unit and to estimate a direction of a sound source associated with the plurality of channels of sound signals using the calculated spatial spectrum. | 03-13-2014 |
20140098972 | ARRAY MICROPHONE DEVICE AND GAIN CONTROL METHOD - The array microphone device has a microphone array composed of a plurality of microphone units, having a signal input section for inputting a signal from the microphone units to be corrected as a signal to be corrected; a reference signal input section; a gain variable section for making the levels of the signal to be corrected and the reference signal equal; and a gain control section. The gain control section includes a high-speed gain update section for changing the gain with a first amount of change per unit time upon time elapsed since the array microphone device is starting-up being below a predetermined period of time; and a low-speed gain update section for changing the gain with a second amount of change per unit time upon the elapsed time being above the predetermined period of time, the second amount of change being smaller than the first amount of change. | 04-10-2014 |
20140105416 | 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. | 04-17-2014 |
20140112496 | MICROPHONE PLACEMENT FOR NOISE CANCELLATION IN VEHICLES - Systems and methods for processing acoustic signals in vehicles are provided. An example system comprises one or more microphones and a voice monitoring device. The voice monitoring device can receive, via the one or more microphones, an acoustic signal and suppress noise in the acoustic signal to obtain a clean speech component. The obtained clean speech component can be provided to one or more vehicle systems. In some embodiments, two microphones selected from the one or more microphones can be positioned on an inner side of a roof of the vehicle, above a windshield, in front of a driver's seat, and directed towards a driver. The two microphones can be equidistant with respect to a symmetry plane of the driver's seat. | 04-24-2014 |
20140119568 | Adaptive Microphone Beamforming - The present invention relates to adaptive beamforming in audio systems. More specifically, aspects of the invention relate to a method for adaptively estimating a target sound signal by establishing a simulation model simulating an audio environment comprising: a plurality of spatially separated microphones, a target sound source, and a number of audio noise sources. | 05-01-2014 |
20140126743 | ACOUSTIC VOICE ACTIVITY DETECTION (AVAD) FOR ELECTRONIC SYSTEMS - Acoustic Voice Activity Detection (AVAD) methods and systems are described. The AVAD methods and systems, including corresponding algorithms or programs, use microphones to generate virtual directional microphones which have very similar noise responses and very dissimilar speech responses. The ratio of the energies of the virtual microphones is then calculated over a given window size and the ratio can then be used with a variety of methods to generate a VAD signal. The virtual microphones can be constructed using either an adaptive or a fixed filter. | 05-08-2014 |
20140126744 | ACOUSTIC VOICE ACTIVITY DETECTION (AVAD) FOR ELECTRONIC SYSTEMS - Acoustic Voice Activity Detection (AVAD) methods and systems are described. The AVAD methods and systems, including corresponding algorithms or programs, use microphones to generate virtual directional microphones which have very similar noise responses and very dissimilar speech responses. The ratio of the energies of the virtual microphones is then calculated over a given window size and the ratio can then be used with a variety of methods to generate a VAD signal. The virtual microphones can be constructed using either an adaptive or a fixed filter. | 05-08-2014 |
20140133674 | AUDIO PROCESSING DEVICE, METHOD AND PROGRAM - An audio processing device including a factorization unit which factorizes frequency information obtained by performing time-frequency transformation on an audio signal of a plurality of channels into a channel matrix representing characteristics of a channel direction, a frequency matrix representing characteristics of a frequency direction, and a time matrix representing characteristics of a time direction; and an extraction unit which extracts the frequency information of audio from an arbitrary designated direction based on the channel matrix, the frequency matrix, and the time matrix. | 05-15-2014 |
20140153740 | BEAMFORMING PRE-PROCESSING FOR SPEAKER LOCALIZATION - Methods and apparatus to beamform a first plurality of microphone signals using at least one beamforming weight to obtain a first beamformed signal, beamform a second plurality of microphone signals using the at least one beamforming weight to obtain a second beamformed signal, and adjust the at least one beam forming weight so that the power density of at least one perturbation component present in the first or the second plurality of microphone signals is reduced. | 06-05-2014 |
20140161277 | COMPRESSOR AUGMENTED ARRAY PROCESSING - The present invention relates generally to the use of compressors, with an optional noise extractor, to improve audio sensing performance of one or more microphones. The audio sensing performance of a single element microphone array with dynamic range compression can be improved by the use of a noise extractor, to modify the operation of the compressor, typically to avoid noise floor amplification. Dynamic range compression can be applied to the output of two or more element microphone array processing with the optional use of a noise extractor. Dynamic range compression can precede the microphone array processing with the optional use of a noise extractor. Syllabic dynamic range compression may be used in one or more element microphone arrays, with the optional use of a noise extractor, which increases speech recognition accuracy. | 06-12-2014 |
20140177867 | SOUND CAPTURE SYSTEM - A sound capture system is disclosed that includes an open-sphere microphone array where at least four omnidirectional microphones providing at least four output signals are disposed around a point of symmetry and an evaluation circuit that is connected to the at least four microphones disposed around the point of symmetry and that is configured to superimpose the output signal of each of the at least four microphones disposed around the point of symmetry with the output signal of one of the other microphones to form at least four differential microphone constellations providing at least four output signals, each differential microphone constellation having an axis along which it exhibits maximum sensitivity. | 06-26-2014 |
20140185823 | IMMERSIVE 3D SOUND SPACE FOR SEARCHING AUDIO - Systems, methods, and computer-readable storage media for generating an immersive three-dimensional sound space for searching audio. The system generates a three-dimensional sound space having a plurality of sound sources playing at a same time, wherein each of the plurality of sound sources is assigned a respective location in the three-dimensional sound space relative to one another, and wherein a user is assigned a current location in the three-dimensional sound space relative to each respective location. Next, the system receives input from the user to navigate to a new location in the three-dimensional sound space. Based on the input, the system then changes each respective location of the plurality of sound sources relative to the new location in the three-dimensional sound space. | 07-03-2014 |
20140185824 | FORMING VIRTUAL MICROPHONE ARRAYS USING DUAL OMNIDIRECTIONAL MICROPHONE ARRAY (DOMA) - A dual omnidirectional microphone array noise suppression is described. Compared to conventional arrays and algorithms, which seek to reduce noise by nulling out noise sources, the array of an embodiment is used to form two distinct virtual directional microphones which are configured to have very similar noise responses and very dissimilar speech responses. The only null formed is one used to remove the speech of the user from V | 07-03-2014 |
20140185825 | FORMING VIRTUAL MICROPHONE ARRAYS USING DUAL OMNIDIRECTIONAL MICROPHONE ARRAY (DOMA) - A dual omnidirectional microphone array noise suppression is described. Compared to conventional arrays and algorithms, which seek to reduce noise by nulling out noise sources, the array of an embodiment is used to form two distinct virtual directional microphones which are configured to have very similar noise responses and very dissimilar speech responses. The only null formed is one used to remove the speech of the user from V | 07-03-2014 |
20140192997 | Sound Collection Method And Electronic Device - A sound collection method and an electronic device are disclosed. The method is applicable to an electronic device that includes an image acquisition unit and an audio collection unit. The method includes determining a focus object when the image acquisition unit is acquiring images; obtaining a position relationship information between the focus object and the image acquisition unit based on the focus object; obtaining a first direction information based on the position relationship information; and controlling the audio collection unit to collect the sound from a sound source corresponding to the first direction based on the first direction information. | 07-10-2014 |
20140192998 | ADVANCED SPEECH ENCODING DUAL MICROPHONE CONFIGURATION (DMC) - A microphone array is described for use in ultra-high acoustical noise environments. The microphone array includes two directional close-talk microphones. The two microphones are separated by a short distance so that one microphone picks up more speech than the other. The microphone array can be used along with an adaptive noise removal program to remove a significant portion of noise from a speech signal of interest. | 07-10-2014 |
20140192999 | METHOD AND APPARATUS FOR LOCALIZATION OF AN ACOUSTIC SOURCE AND ACOUSTIC BEAMFORMING - Embodiments include a method and an apparatus for the localization of at least one source of an acoustic signal including: temporally sampling the acoustic signal with a plurality of microphones to obtain a (D+1)-dimensional space-time matrix representation of the acoustic signal, wherein D is the number of spatial dimensions, applying a (D+1)-dimensional Fourier transform to the matrix representation, determining a first peak in a spectrum obtained based on the application of the Fourier transform, and calculating the direction of arrival of the acoustic signal at at least one of the plurality of microphones based on the determined first peak. | 07-10-2014 |
20140219471 | USER VOICE LOCATION ESTIMATION FOR ADJUSTING PORTABLE DEVICE BEAMFORMING SETTINGS - An audio device may use the audio detected at two opposite facing, front and rear omnidirectional microphones to determine the angular directional location of a user's voice while the device in speaker mode or audio command input mode. The angular directional location may be determined to be at front, side and rear locations of the device during the period of time by calculating an energy ratio of audio signals output by the front and rear microphones during the period. Comparing the ratio to experimental data for sound received from different directions around the device may provide the location of the user's voice. Based on the determination, audio beamforming input settings may be adjusted for user voice beamforming. As a result, the device can perform better beamforming to combine the signals captured by the microphones and generate a single output that isolates the user's voice from background noise. | 08-07-2014 |
20140219472 | SOUND COLLECTING SYSTEM AND ASSOCIATED METHOD - A sound collecting system includes a plurality of microphones, a distance estimation module and an adjustment module. The distance estimation module estimates a distance to a user to accordingly provide a user distance. The adjustment module adjusts a part or all of the positions of the microphones according to the user distance. | 08-07-2014 |
20140233757 | Noisy Environment Communication Enhancement System - A communication system enhances communications in a noisy environment. Multiple input arrays convert a voiced or unvoiced signal into an analog signal. A converter receives the analog signal and generates digital signals. A digital signal processor determines temporal and spatial information from the digital signals. The processed signals are then converted to audible sound. | 08-21-2014 |
20140241549 | Robust Estimation of Sound Source Localization - A method for sound source localization in a digital system having at least two audio capture devices is provided that includes receiving audio signals from the two audio capture devices, computing a signal-to-noise ratio (SNR) for each frequency band of a plurality of frequency bands in a processing frame of the audio signals, determining a frequency band weight for each frequency band of the plurality of frequency bands based on the SNR computed for the frequency band, computing an estimated time delay of arrival (TDOA) of sound for the processing frame using the frequency band weights, and converting the estimated TDOA to an angle representing sound direction. | 08-28-2014 |
20140247953 | SPEAKER LOCALIZATION - Methods and apparatus for determining phase shift information between the first and second microphone signals for a sound signal, and determining an angle of incidence of the sound in relation to the first and second positions of the first and second microphones from the phase shift information of a band-limited test signal received by the first and second microphones for a frequency range of interest. | 09-04-2014 |
20140247954 | Entrained Microphones - In some embodiments, a microphone system may include a deformable element that may be made of a material that is subject to deformation in response to external phenomenon. Sensing ports may be in contact with a respective region of the deformable element and may be configured to sense a deformation of a region of the deformable element and generate a signal in response thereto. The plurality of signals may be useable to determine spatial dependencies of the external phenomenon. The external phenomenon may be pressure and the signals may be useable to determine spatial dependencies of the pressure. | 09-04-2014 |
20140254823 | Distributed Automatic Level Control for a Microphone Array - A distributed automatic level control function is provided, in which information relating to a common automatic level control parameter is transmitted to each of a plurality of microphone devices, wherein the information transmitted to at least one microphone device is derived from an audio sample of at least one different microphone device. Each microphone device produces the common automatic level control parameter based on the information received by the microphone device and applies the common automatic level control parameter produced by the microphone device to a distributed automatic level controller of the microphone device. | 09-11-2014 |
20140254824 | MICROPHONE APPARATUS - A microphone array disposed on a two-piece computer is provided. The two-piece computer has a first piece with a keyboard disposed thereon and a second piece with a display screen disposed thereon. The microphone array has two microphones, which are both disposed on the first piece, wherein a connecting line of the two microphones is not vertical to a normal line of the display screen of the second piece. | 09-11-2014 |
20140270245 | POLYHEDRAL AUDIO SYSTEM BASED ON AT LEAST SECOND-ORDER EIGENBEAMS - A microphone array-based audio system that supports representations of auditory scenes using second-order (or higher) harmonic expansions based on the audio signals generated by the microphone array. In one embodiment, a plurality of audio sensors are mounted on the surface of an acoustically rigid polyhedron that approximates a sphere. The number and location of the audio sensors on the polyhedron are designed to enable the audio signals generated by those sensors to be decomposed into a set of eigenbeams having at least one eigenbeam of order two (or higher). Beamforming (e.g., steering, weighting, and summing) can then be applied to the resulting eigenbeam outputs to generate one or more channels of audio signals that can be utilized to accurately render an auditory scene. | 09-18-2014 |
20140270246 | INTERFACE FOR A DIGITAL MICROPHONE ARRAY - An interface for an array of digital microphones in an electronic device may include a head-end chip coupled to the digital microphones through a bus. The bus may be shared by each microphone of the array of microphones and be multiplexed to allow transmission of data from the microphones to the head-end chip and transmission of power from the head-end chip to the array of digital microphones. The head-end chip may perform signal processing on receive data from the array of digital microphones to create beamforming arrays. The array of microphones may include microphones with different characteristics to improve performance of the array of microphones. | 09-18-2014 |
20140270247 | BEAMFORMING A DIGITAL MICROPHONE ARRAY ON A COMMON PLATFORM - An interface for an array of digital microphones in an electronic device may include a head-end chip coupled to the digital microphones through a bus. The bus may be shared by each microphone of the array of microphones and be multiplexed to allow transmission of data from the microphones to the head-end chip and transmission of power from the head-end chip to the array of digital microphones. The head-end chip may perform signal processing on receive data from the array of digital microphones to create beamforming arrays. The array of microphones may include microphones with different characteristics to improve performance of the array of microphones. | 09-18-2014 |
20140270248 | Method and Apparatus for Detecting and Controlling the Orientation of a Virtual Microphone - A method for controlling the orientation of a virtual microphone, which is carried out on an electronic device, includes combining and processing signals from a microphone array to create a virtual microphone; receiving data from a sensor of the electronic device; determining, based on the received data, a mode in which the electronic device is being used; and based on the determined mode, directionally orienting the virtual microphone. Possible use modes include a) a stowed use mode, in which the criterion is the electronic device being substantially enclosed by surrounding material; b) a handset (alternately, private) use mode, in which the criterion is the electronic device being held proximate to a user; and c) a handheld (alternately, speakerphone) use mode, in which the criterion is the electronic device being held away from a user. | 09-18-2014 |
20140294196 | USER DEVICE HAVING PLURALITY OF MICROPHONES AND OPERATING METHOD THEREOF - A user device having a plurality of microphones and an operating method thereof are provided. The user device includes a plurality of microphones for converting sounds into electrical signals, an analysis unit for analyzing acoustic characteristics of the electrical signals outputted from the plurality of microphones, a switch unit for electrically connecting a specific microphone and a host device according to the analysis result of the analysis unit, and the host device for executing a function using an electrical signal outputted from the specific microphone. Various other exemplary embodiments are possible. | 10-02-2014 |
20140294197 | Sound Discrimination Method and Apparatus - A method of distinguishing sound sources includes the step of transforming data, collected by at least two transducers which each react to a characteristic of an acoustic wave, into signals for each transducer location. The transducers are separated by a distance of less than about 70 mm or greater than about 90 mm. The signals are separated into a plurality of frequency bands for each transducer location. For each band a comparison is made of the relationship of the magnitudes of the signals for the transducer locations with a threshold value. A relative gain change is caused between those frequency bands whose magnitude relationship falls on one side of the threshold value and those frequency bands whose magnitude relationship falls on the other side of the threshold value. As such, sound sources are discriminated from each other based on their distance from the transducers. | 10-02-2014 |
20140307894 | METHOD AND APPARATUS FOR PROCESSING SIGNALS OF A SPHERICAL MICROPHONE ARRAY ON A RIGID SPHERE USED FOR GENERATING AN AMBISONICS REPRESENTATION OF THE SOUND FIELD - Spherical microphone arrays capture a three-dimensional sound field (P(Ω | 10-16-2014 |
20140314250 | POSITION ESTIMATION SYSTEM USING AN AUDIO-EMBEDDED TIME-SYNCHRONIZATION SIGNAL AND POSITION ESTIMATION METHOD USING THE SYSTEM - Disclosed is a method of estimating a position by a position estimation system. The method includes generating a time-synchronization signal for position determination using a genetic algorithm, embedding the generated time-synchronization signal in an audio signal, and replaying the audio signal embedded with the time-synchronization signal through a speaker, receiving the audio signal embedded with the time-synchronization signal in a microphone, calculating a time delay value of the time-synchronization signal embedded in the received audio signal, and estimating a position of the microphone based on the calculated time delay value. | 10-23-2014 |
20140314251 | BROADBAND SENSOR LOCATION SELECTION USING CONVEX OPTIMIZATION IN VERY LARGE SCALE ARRAYS - Systems and methods are provided to determine a subset of D microphones in a set of N microphones on a perimeter of a space to monitor a target location. The space is divided into L interference locations. An equation is solved to determine microphone weights for the N microphones by minimizing the maximum gain for signals related to the target location and interference locations, further optimized over an l | 10-23-2014 |
20140321664 | METHODS FOR DYNAMICALLY PROGRAMMING A MICROPHONE - Methods for dynamically programming a microphone are provided. The method, adopted by a microphone system including a first microphone device and a host device connected thereto, includes: transmitting, by the host device, a command message to the first microphone device; receiving, by the first microphone device, a command message from the host device; decoding, by the first microphone device, the command message; dynamically performing, by the first microphone device, an operation based on the decoded command message to generate first data; and receiving, by the host device, first data from the first microphone device. | 10-30-2014 |
20140321665 | SOUND PROCESSING DEVICE, AND SOUND PROCESSING METHOD - A sound processing apparatus ( | 10-30-2014 |
20140328496 | CALIBRATED DUAL OMNIDIRECTIONAL MICROPHONE ARRAY (DOMA) - Systems and methods are described by which microphones are calibrated. Disclosed are techniques for generating a first output signal from a first input signal at a first microphone, generating a second output signal from a second input signal at a second microphone, forming a first filter as a function of the first output signal and the second output signal, the first filter being configured to substantially model the first microphone, and forming a second filter as a function of the first output signal and the second output signal, the second filter being configured to substantially model the second microphone. The second filter may be used to output a third output signal from the first output signal, and the first filter may be used to output a fourth output signal from the second output signal. The fourth output signal may be substantially similar to the third output signal. | 11-06-2014 |
20140328497 | CALIBRATED DUAL OMNIDIRECTIONAL MICROPHONE ARRAY (DOMA) - Systems and methods are described by which microphones are calibrated. Disclosed are techniques for generating a first output signal from a first input signal at a first microphone, generating a second output signal from a second input signal at a second microphone, forming a first filter as a function of the first output signal and the second output signal, the first filter being configured to substantially model the first microphone, and forming a second filter as a function of the first output signal and the second output signal, the second filter being configured to substantially model the second microphone. The second filter may be used to output a third output signal from the first output signal, and the first filter may be used to output a fourth output signal from the second output signal. A first virtual microphone and a second virtual microphone may be formed using the third output signal and the fourth output signal. | 11-06-2014 |
20140334639 | DIRECTIVITY CONTROL METHOD AND DEVICE - Directivity control method and device which can emphasize or suppress sound deriving from an arbitrary direction with a little computation using two microphones closely disposed are provided. An interchange circuit | 11-13-2014 |
20140341392 | AUGMENTATION OF A BEAMFORMING MICROPHONE ARRAY WITH NON-BEAMFORMING MICROPHONES - Embodiments of the present disclosure include an apparatus ( | 11-20-2014 |
20140348342 | AUDIO LENS - An apparatus configured to: determine a viewing angle associated with at least one apparatus camera; determine from at least two audio signals at least one audio source orientation relative to an apparatus; and generate at least one spatial filter including at least a first orientation range associated with the viewing angle and a second orientation range relative to the apparatus. | 11-27-2014 |
20140355784 | DIRECTIONAL MICROPHONE AND OPERATING METHOD THEREOF - A directional microphone and an operating method thereof include a first signal generator generating a first sound signal corresponding to a front sound coming through a front sound hole of the directional microphone. A second signal generator generates a second sound signal corresponding to a rear sound coming through a rear sound hole of the directional microphone. A phase delay controller delays a phase of the rear sound coming through the rear sound hole, and a signal processor synthesizes the first sound signal and the second sound signal. | 12-04-2014 |
20140355785 | MOBILE DEVICE LOCALIZATION USING AUDIO SIGNALS - Mobile device localization using audio signals is described. In an example, a mobile device is localized by receiving a first audio signal captured by a microphone located at the mobile device and a second audio signal captured from a further microphone. A correlation value between the first audio signal and second audio signal is computed, and this is used to determine whether the mobile device is in proximity to the further microphone. In one example, the mobile device can receive the audio signals from the further microphone and calculate the correlation value. In another example, a server can receive the audio signals from the mobile device and the further microphone and calculate the correlation value. In examples, the further microphone can be a fixed microphone at a predetermined location, or the further microphone can be a microphone located in another mobile device. | 12-04-2014 |
20140376740 | DIRECTIVITY CONTROL SYSTEM AND SOUND OUTPUT CONTROL METHOD - A system includes an imaging part that captures an image, a sound collection part that collects sounds, a display part that displays image data captured by the imaging part, a directive direction calculation part that calculates a directive direction which directs toward a sound position corresponding to a designated position of the image data from the sound collection part when any position of the displayed image data is designated, and a control part that forms a directivity in the sounds in the calculated directive direction. The control part controls output of the sounds collected by the sound collection part or output of the sounds which are collected by the sound collection part and of which the directivity is formed, or suspends collection of sounds in the sound collection part, when it is determined that the sound position is included in a preset protection region. | 12-25-2014 |
20140376741 | AUDIO SOURCE POSITION ESTIMATION - An apparatus for determining a position estimate for an audio source comprises two microphones (M | 12-25-2014 |
20150016628 | METHOD AND CIRCUITRY FOR DIRECTION OF ARRIVAL ESTIMATION USING MICROPHONE ARRAY WITH A SHARP NULL - A device is configured for identifying a direction of a sound. The device includes a controller comprising circuitry. The circuitry is configured to receive a first output from a first input device and a second output from a second input device. The circuitry is also configured to add a delay to the second output. The circuitry is also configured to compare the first output to the delayed second output in a plurality of directions to form a comparison. The circuitry is also configured to identify a number of null directions of the plurality of directions where a set of nulls exists based on the comparison. | 01-15-2015 |
20150016629 | DIRECTIONAL MICROPHONE DEVICE, ACOUSTIC SIGNAL PROCESSING METHOD, AND PROGRAM - A directional microphone includes: a microphone which generates a first acoustic signal that has sensitivity in a target direction; a microphone which generates a second acoustic signal that has a blind spot in sensitivity in the target direction; a correction unit which multiplies, in a frequency domain, the second acoustic signal by the first acoustic signal N times, where N is greater than zero, to generate a third acoustic signal which includes the second acoustic signal that has a narrowed angular range of the blind spot in sensitivity in the target direction; and a suppression unit which performs noise suppression using the first acoustic signal as a main signal and the third acoustic signal generated as a reference signal, to generate an output acoustic signal which is the first acoustic signal that has narrowed directivity in the target direction. | 01-15-2015 |
20150023524 | DIRECTIVITY CONTROL SYSTEM, DIRECTIVITY CONTROL METHOD, SOUND COLLECTION SYSTEM AND SOUND COLLECTION CONTROL METHOD - A directivity control system includes: a sound collection unit, configured to collect a sound; a display unit, configured to display a designation screen used to designate a directivity direction oriented from the sound collection unit to a first sound position; a directivity direction calculation unit, configured to calculate a horizontal angle and a vertical angle from the sound collection unit to the first sound position corresponding to the designated directivity direction in accordance with a designation of the directivity direction on the designation screen displayed by the display unit; and a control unit, configured to form directivity of the sound collected by the sound collection unit based on the horizontal angle and the vertical angle calculated by the directivity direction calculation unit. | 01-22-2015 |
20150030179 | PRESERVING PHASE SHIFT IN SPATIAL FILTERING - For preserving phase shift in spatial filtering is disclosed, an electronic device includes a microphone array. A filtering module spatially filters a plurality of received audio signals from the microphone array to increase the signal-to-noise ratio in one or more corresponding output audio signals. A phase module preserves a phase shift of at least one received audio signal in the corresponding output audio signal. | 01-29-2015 |
20150049878 | DIRECTIONAL MEMS MICROPHONE - The present invention relates to a directional MEMS microphone which comprises a cover, a printed circuit board, a sound inlet structure, an integrated circuit chip which is attached to the printed circuit board, and a MEMS die which is attached to the printed circuit board. The cover provides an open side where the printed circuit board and the cover are coupled. The sound inlet structure comprises a main sound port which is disposed on the printed circuit board and communicated with the inner cavity of the MEMS die and a secondary sound port which is communicated with the inner cavity of the cover. The MEMS microphone of the present invention features with directional receiving function, simple structure and convenient application. | 02-19-2015 |
20150055795 | Microphone, A Microphone Arrangement and a Method for Processing Signals in a Microphone - A microphone includes a first signal input, a second signal input and a control unit coupled to the first signal input and the second signal input. The control unit is configured to, upon receiving a support signal at one of the first signal input and second signal input, process a signal received at the other of the first signal input and second signal input as an incoming data signal. | 02-26-2015 |
20150055796 | 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. | 02-26-2015 |
20150055797 | METHOD AND DEVICE FOR LOCALIZING SOUND SOURCES PLACED WITHIN A SOUND ENVIRONMENT COMPRISING AMBIENT NOISE - A method for localizing one or more sound sources of interest placed within a sound environment comprising ambient noise by estimating the directions of arrival (θ,φ) of said one or more sound source of interest comprising the steps of: calculating the environment steered response power (SRP (t, f, θ, φ)) corresponding to the steered response power of said one or more source of interest for one or more orientations using said environment audio signals; obtaining, using said array of at least two microphones, noise audio signals corresponding to the audio signals emanating from said sound environment under particular reference conditions; calculating a noise steered response power (SRP | 02-26-2015 |
20150055798 | METHOD FOR VOICE RECORDING AND ELECTRONIC DEVICE THEREOF - A method for voice recording and an electronic device thereof are provided. The method includes determining a voice recording mode from among a plurality of voice recording modes, determining a voice beamforming direction according to the determined voice recording mode, and recording voice signals based on the determined voice beamforming direction. | 02-26-2015 |
20150055799 | Synchronization of Buffered Data in Multiple Microphones - First analog signals are received from a first microphone, converted into first digital data and stored in a first buffer. A determination is made as to whether voice activity has occurred when voice activity is determined, a voice activity detect signal is sent to an external processor. The external processor responsively provides an exterior clock signal upon receiving the voice activity detect signal. Second analog signals are received from a second microphone, converted into second digital data and stored in a second buffer. The first digital data in the first buffer is not necessarily synchronized in real time with the second digital data in the second buffer. The first digital data from the first buffer and the second digital data from the second buffer is decimated using the external clock to provide decimated output data, the decimated output data having the first digital data and the second digital data aligned in real time. | 02-26-2015 |
20150063589 | METHOD, APPARATUS, AND MANUFACTURE OF ADAPTIVE NULL BEAMFORMING FOR A TWO-MICROPHONE ARRAY - A method, apparatus, and manufacture of beamforming is provided. Adaptive null beamforming is performed for signals from first and second microphones of a two-microphone array. The signals from the microphones are decomposed into subbands. Beamforming weights are evaluated and adaptively updated over time based, at least in part, on the direction of arrival and distance of the target signal. The beamforming weights are applied to the subbands at each updated time interval. Each subband is then combined. | 03-05-2015 |
20150063590 | SOUND SOURCE SEPARATING APPARATUS, SOUND SOURCE SEPARATING PROGRAM, SOUND PICKUP APPARATUS, AND SOUND PICKUP PROGRAM - There is provided a sound source separating apparatus including a bidirectionality forming unit configured to form a bidirectionality by use of a sound signal picked up by two microphones which are located to be horizontal with respect to the target direction, among three microphones disposed at vertexes of an isosceles right triangle, a unidirectionality forming unit configured to form a unidirectionality by use of a sound signal picked up by two microphones which are located in a same direction as the target direction, among the three microphones, and a target sound extracting unit configured to extract a target sound by performing a spectral subtraction of all outputs from the bidirectionality forming unit and the unidirectionality forming unit from a signal. | 03-05-2015 |
20150063591 | SOUND RECEIVING SYSTEM - A sound receiving system is disclosed, each of the plurality of basic array devices has an output terminal connected with one filter, each of the plurality of filters has an output terminal connected with an input terminal of the second sound-mixing output device; the basic array device includes a microphone array, the microphone array includes a plurality of microphones longitudinally arranged along a straight line in order, and two adjacent microphones in the microphone array are separated with a distance of | 03-05-2015 |
20150071460 | 2-Way Enhanced Live Recording Splicing (ELRS) - A method of live audio recording intended to enhance the playback performance of vocal, conventional, analog and digital equipment by using both microphones and direct current to recording apparatuses to capture the performance, splicing together two or more recordings and producing them through the use of volume adjustment and/or separate editing equipment or software. | 03-12-2015 |
20150078581 | Systems And Methods For Audio Conferencing - Systems and methods for enabling an audio conference are provided. In one aspect, a system includes a transmitting device which receives audio signals representing sounds captured by at least two microphones from participants situated at a first location. A time-of-arrival delay between at least two of the audio signals is calculated and a beam-formed monaural audio signal and corresponding spatial data are generated and transmitted to remote participants situated at a second location. A receiving device processes the beam-formed monaural audio signal based on the spatial data to render and output spatial audio via speakers to the remote participants at the second location. In various aspects, the spatial data may include an angular value or a participant identifier that are determined based on the time-of-arrival delay. The spatial data may also indicate a total number of conference participants that are detected at the first location. | 03-19-2015 |
20150078582 | Beamforming Microphone Array with Support for Interior Design Elements - Embodiments of the present disclosure include an apparatus ( | 03-19-2015 |
20150086037 | SOUND RECEIVING DEVICE - A sound receiving device is disclosed, including a microphone array, a plurality of time delay circuits and a sound-mixing output device, wherein the microphone array includes a plurality of microphones longitudinally arranged along a straight line in order, two adjacent microphones in the microphone array are separated with a distance of | 03-26-2015 |
20150086038 | TIME-FREQUENCY DIRECTIONAL PROCESSING OF AUDIO SIGNALS - An approach to processing of acoustic signals acquired at a user's device include one or both of acquisition of parallel signals from a set of closely spaced microphones, and use of a multi-tier computing approach in which some processing is performed at the user's device and further processing is performed at one or more server computers in communication with the user's device. The acquired signals are processed using time versus frequency estimates of both energy content as well as direction of arrival. In some examples, a non-negative matrix or tensor factorization approach is used to identify multiple sources each associated with a corresponding direction of arrival of a signal from that source. In some examples, data characterizing direction of arrival information is passed from the user's device to a server computer where direction-based processing is performed. | 03-26-2015 |
20150086039 | Wearable Directional Microphone Array Apparatus and System - 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. | 03-26-2015 |
20150092958 | DEVICES AND METHODS FOR AUDIBLE INDICATORS EMANATING FROM SELECTED LOCATIONS - Methods, systems, and devices are described for providing audio to one or more individuals in an operating room. An ultrasonic signal generator may be provided that generates two or more ultrasonic signals that combine to produce an audible signal at a desired location. The audio signal may be perceived by individuals in the operating room to emanate from a surface or location within the operating room, or the audio signal may be generated to provide an audible signal to one or more persons within a particular location within the operating room. Multiple audio signals may be generated to emanate from multiple different locations. Likewise, multiple audio signals may be generated to provide different audible signals in different locations in the operating room. | 04-02-2015 |
20150110288 | AUGMENTED ELLIPTICAL MICROPHONE ARRAY - In one embodiment, an audio system has a microphone array and a signal processing subsystem that processes audio signals generated by the microphone array to produce an output beampattern. The microphone array has (i) a first microphone set of two or more microphones located on a first ellipse, (ii) a second microphone set of two or more microphones located on a second ellipse within the first ellipse, and (iii) a third microphone set of one or more microphones located within the second ellipse, where the microphones in the first, second, and third microphone sets are effectively all in one plane. The signal processing subsystem has (1) a decomposer that spatially decomposes the microphone audio signals to generate a plurality of eigenbeams and (2) a beamformer that generates the output beampattern as a weighted sum of the eigenbeams. | 04-23-2015 |
20150117671 | METHOD AND APPARATUS FOR CALIBRATING MULTIPLE MICROPHONES - In one embodiment, a method includes capturing sound using a plurality of microphones, wherein the plurality of microphones is associated with a computing system. The method also includes determining energy levels for the plurality of microphones, and determining signal-to-noise ratios (SNRs) for the plurality of microphones. Finally, the method includes selecting a particular microphone of the plurality of microphones based on the energy levels and the SNRs, wherein selecting the particular microphone includes providing audio signals obtained by the particular microphone to the computing system for use. | 04-30-2015 |
20150117672 | MICROPHONE ARRAY - A spherical microphone array that includes a sound-diffracting structure having a closed three-dimensional shape of at least one non-regular, regular or semi-regular convex polyhedron with congruent faces of regular or non-regular polygons and at least two omnidirectional microphones disposed in or on the sound-diffracting structure on an oval line whose center is disposed on a center line that subtends the center of one of the faces of the regular polygons. | 04-30-2015 |
20150117673 | DIGITAL SIGNAL PROCESSING WITH ACOUSTIC ARRAYS - Methods, systems, and techniques of digital signal processing using acoustic arrays are provided. Example embodiments described herein provide enhanced acoustic arrays that utilize MEMS digital microphones to offer greater control and measurement capabilities to users and systems that desire to measure sound typically to derive other data. Large numbers of digital microphones can be manufactured to be placed on an acoustic array to derive a plurality of derived acoustic array measurements. | 04-30-2015 |
20150124997 | SOUND PROCESSING DEVICE, AND SOUND PROCESSING METHOD - A sound processing apparatus ( | 05-07-2015 |
20150139444 | SYSTEMS AND METHODS FOR DETECTING TRANSIENT ACOUSTIC SIGNALS - A two-scale array for detecting wind noise signals and acoustic signals includes a plurality of subarrays each including a plurality of microphones. The subarrays are spaced apart from one another such that the subarrays are configured to detect acoustic signals, and the plurality of microphones in each subarray are located close enough to one another such that wind noise signals are substantially correlated between the microphones in each subarray. | 05-21-2015 |
20150146882 | MICROPHONE ARRAY SYSTEM AND A METHOD FOR SOUND ACQUISITION - A microphone array system for sound acquisition from multiple sound sources in a reception space surrounding a microphone array that is interfaced with a beamformer module is disclosed. The microphone array includes microphone transducers that are arranged relative to each other in N-fold rotationally symmetry, and the beamformer includes beamformer weights that are associated with one of a plurality of spatial reception sectors corresponding to the N-fold rotational symmetry of the microphone array. Microphone indexes of the microphone transducers are arithmetically displaceable angularly about the vertical axis during a process cycle, so that a same set of beamformer weights is used selectively for calculating a beamformer output signal associated with any one of the spatial reception sectors. A sound source location module is also disclosed that includes a modified steered power response sound source location method. A post filter module for a microphone array system is also disclosed. | 05-28-2015 |
20150146883 | RESPONSE-COMPENSATED MICROPHONE - Systems and methods are disclosed for managing input to and output from a microphone, including adapting the microphone's response to changing polar response patterns among multiple microphone capsules, providing output via multi-colored lights to reflect the system state and operational characteristics, and sending various information to and from the microphone (such as carrying power to the microphone, digital and/or analog audio from the microphone, and data to and/or from the microphone) via a single cable. | 05-28-2015 |
20150296289 | APPARATUS AND METHOD FOR ENHANCING AN AUDIO OUTPUT FROM A TARGET SOURCE - A computer-program product embodied in a non-transitory computer read-able medium that is programmed for transmitting audio data to at least one output for audio playback. The computer-program product comprises instructions for receiving at least one of a digital image of a target source using a camera, and distance and angle information of the target source entered at a user interface. The computer-program product comprises instructions for generating one or more first coordinates based on the at least one of the digital image and the distance and angle information. The computer-program product comprises instructions for adjusting a sensitivity of a first microphone based on the one or more first coordinates. The computer-program product comprises instructions for receiving audio data from the target source in response to adjusting the sensitivity of the first microphone and transmitting the audio data to one or more outputs for audio playback. | 10-15-2015 |
20150304765 | Method and System for Obtaining an Audio Signal - A method and system for obtaining an audio signal. In one embodiment, the method comprises receiving a first sound signal at a first microphone arranged at a first height vertically above a substantially flat surface; receiving a second sound signal at a second microphone arranged at a second height vertically above the substantially flat surface; processing a signal provided by the first microphone using a low pass filter; processing a signal provided by the second microphone using a high pass filter; adding the signals processed by the low pass filter and the high pass filter to form a sum signal; and outputting the sum signal as an audio signal. | 10-22-2015 |
20150304766 | METHOD FOR SPATIAL FILTERING OF AT LEAST ONE SOUND SIGNAL, COMPUTER READABLE STORAGE MEDIUM AND SPATIAL FILTERING SYSTEM BASED ON CROSS-PATTERN COHERENCE - Method for spatial filtering of at least one sound signal (M | 10-22-2015 |
20150312662 | SOUND PROCESSING APPARATUS, SOUND PROCESSING SYSTEM AND SOUND PROCESSING METHOD - A sound processing apparatus includes a processor to obtain sound data and image data, wherein the sound data is collected from a sound source in a given area by a sound collection unit including a plurality of microphones and the image data is captured by an imaging unit which captures an image at least partially in the given area, to designate a direction defined relative to the sound collection unit, wherein the designated direction corresponds to a designation part on an image displayed based on the image data, to designate an arbitrary range in the given area, wherein the designated arbitrary range corresponds to a designation part on the image displayed based on the image data, and to emphasize a sound component in the sound data in the direction designated by the first designation unit within the arbitrary range designated by the second designation unit. | 10-29-2015 |
20150312663 | SOURCE SEPARATION USING A CIRCULAR MODEL - An approach to separating multiple sources exploits the observation that each source is associated with a linear-circular phase characteristic in which the relative phase between pairs of microphones follows a linear (modulo) pattern. In some examples, a modified RANSAC (Random Sample Consensus) approach is used to identify the frequency/phase samples that are attributed to each source. In some examples, either in combination with the modified RANSAC approach or using other approaches, a wrapped variable representation is used to represent a probability density of phase, thereby avoiding a need to “unwrap” phase in applying probabilistic techniques to estimating delay between sources. | 10-29-2015 |
20150319524 | APPARATUS AND METHOD FOR DETECTING LOCATION OF MOVING BODY, LIGHTING APPARATUS, AIR CONDITIONING APPARATUS, SECURITY APPARATUS, AND PARKING LOT MANAGEMENT APPARATUS - An apparatus and method for detecting the location of a moving body. The apparatus may include a speaker configured to intermittently generate a pulse signal to a detection space and a microphone array configured to obtain a reflected sound of the pulse signal that is generated in the detection space. The location of a moving body is estimated by extracting a change of a sound field from the reflected sound. The present invention can be applied to an apparatus for detecting a moving body without any side effect because the location of a moving body can be accurately detected up to an angle and distance. Furthermore, the present invention can be applied to various application apparatuses, such as a lighting apparatus, an air conditioning apparatus, a security apparatus, and a parking lot management apparatus because there is no problem in health and there is no side effect in application. | 11-05-2015 |
20150326968 | DIRECTIVITY CONTROL APPARATUS, DIRECTIVITY CONTROL METHOD, STORAGE MEDIUM AND DIRECTIVITY CONTROL SYSTEM - A directivity control apparatus controls a directivity of a sound collected by a first sound collecting unit including a plurality of microphones. The directivity control apparatus includes a directivity forming unit, configured to form a directivity of the sound in a direction toward a monitoring target corresponding to a first designated position in an image displayed on a display unit, and an information obtaining unit, configured to obtain information on a second designated position in the image displayed on the display unit, designated in accordance with a movement of the monitoring target. The directivity forming unit is configured to change the directivity of the sound toward the monitoring target corresponding to the second designated position by referring to the information on the second designated position obtained by the information obtaining unit. | 11-12-2015 |
20150341719 | Precise Tracking of Sound Angle of Arrival at a Microphone Array under Air Temperature Variation - A video conference endpoint detects a face and determines a face angle of the detected face relative to a reference direction based on images captured with a camera. The endpoint determines an angle of arrival of sound (i.e., a sound angle) received at a microphone array that transduces the sound relative to the reference direction based on the transduced sound and a sound speed parameter indicative of a speed of sound in air. The endpoint compares the face angle against the sound angle, and adjusts the sound speed parameter so as to reduce the angle difference if the compare indicates an angle difference greater than zero between the face and sound angles. | 11-26-2015 |
20150341720 | VARIABLE DIRECTIVITY ELECTRET CONDENSER MICROPHONE - Provided is a variable directivity electret condenser microphone that can simplify a circuit configuration, and outputs an audio signal in a balanced manner, which includes electrically independent first and second electret condenser microphone units in which first and second fixed electrodes are arranged back to back and facing each other in a mutually non-conductive state, and first and second diaphragms are arranged facing the first and second fixed electrodes with fixed intervals therefrom respectively, a first impedance converter having an input terminal connected to the first fixed electrode, and a first buffer circuit connected to the first impedance converter, a second impedance converter having an input terminal connected to the second fixed electrode, and a second buffer circuit selectively connected to the second impedance converter, and a directivity variable switch that can alternatively select a mode from at least a first directivity mode to a third directivity mode. | 11-26-2015 |
20150341721 | VARIABLE DIRECTIVITY ELECTRET CONDENSER MICROPHONE - Provided is a variable directivity electret condenser microphone that can simplify a circuit configuration, and outputs an audio signal in an unbalanced manner. Included are electrically independent first and second electret condenser microphone units in which first and second fixed electrodes are arranged back to back and facing each other in a mutually non-conductive state, and first and second diaphragms are arranged facing the first and second fixed electrodes with fixed intervals from the first and second fixed electrodes, respectively, a first impedance converter having an input terminal connected to the first fixed electrode, a DC cut capacitor selectively connected between an output terminal of the first impedance converter and an input terminal of the second impedance converter, and a directivity variable switche that can alternatively select a mode from at least a first directivity mode to a third directivity mode. | 11-26-2015 |
20150341723 | MULTITASK LEARNING METHOD FOR BROADBAND SOURCE-LOCATION MAPPING OF ACOUSTIC SOURCES - A method involves acoustic source localization by capitalizing on the sparse nature of a source location map (SLM). Sparsity arises naturally since one seeks the location of K sources in a grid of G tentative locations where G>>K. The source localization problem is cast as a regularized LS regression problem with a sparsity constraint whose solution yields the SLM. An iterative solver based on block coordinate descent (BCD) is used. BCD leads to scalar closed-form updates rendering the method's computational complexity per iteration linear with respect to the grid size. The disclosed method enables high resolution location estimation with fewer array measurements than classical matched field processing methods. | 11-26-2015 |
20150341734 | 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. | 11-26-2015 |
20150358722 | MICROPHONE ARRANGEMENT WITH IMPROVED DIRECTIONAL CHARACTERISTIC - A microphone arrangement with improved directional characteristics is provided with at least two microphones ( | 12-10-2015 |
20150358732 | ADAPTIVE MICROPHONE BEAMFORMING - The present invention relates to adaptive beamforming in audio systems. More specifically, aspects of the invention relate to a method for adaptively estimating a target sound signal by establishing a simulation model simulating an audio environment comprising: a plurality of spatially separated microphones, a target sound source, and a number of audio noise sources. | 12-10-2015 |
20160014490 | APPARATUS, METHOD AND SYSTEM OF COMMUNICATING ACOUSTIC INFORMATION OF A DISTRIBUTED MICROPHONE ARRAY BETWEEN MOBILE DEVICES | 01-14-2016 |
20160014506 | MICROPHONE ARRAY CONTROL APPARATUS AND MICROPHONE ARRAY SYSTEM | 01-14-2016 |
20160029117 | Apparatus - Apparatus including: an acoustic transducer, and a sound channel coupled to the acoustic transducer, the sound channel including an element having a shape that is electrically controllable, wherein the shape of the element is electrically controllable to change the acoustic properties of the sound channel. | 01-28-2016 |
20160029122 | MULTIPLE DEVICE NOISE REDUCTION MICROPHONE ARRAY - Various embodiments are directed to cooperation among communications devices having microphones to employ their microphones in unison to provide voice detection with noise reduction for voice communications. A first communications device comprises a processor circuit; a first microphone; an interface operative to communicatively couple the processor circuit to a network; and a storage communicatively coupled to the processor circuit and arranged to store a sequence of instructions operative on the processor circuit to store a first detected data that represents sounds detected by the first microphone; receive a second detected data via the network that represents sounds detected by a second microphone of a second communications device; subtractively sum the first and second data to create a processed data; and transmit the processed data to a third communications device. Other embodiments are described and claimed herein. | 01-28-2016 |
20160037254 | Unidirectional Close-Talking Microphone and Microphone Cap - The orientation of the directional axis and the directionality of a microphone are adjusted through a simple configuration. A unidirectional close-talking microphone includes a microphone unit including a front sound-collecting segment and a rear sound-collecting segment; and a microphone cap attachable to the outer circumference of the microphone cap, the microphone cap including a plurality of sound-collecting holes on a side face, the relative position between the microphone cap and the microphone unit being switchable between a first position and a second position along the central axis, the sound-collecting holes being disposed on opposite sides of the central axis at different positions along the central axis, the rear sound-collecting segment being in communication with outside of the microphone cap through the sound-collecting holes in the microphone cap when the microphone cap resides at the first position, part of the rear sound-collecting segment being covered with the microphone cap when the microphone cap resides at the second position. | 02-04-2016 |
20160044408 | BOUNDARY MICROPHONE AND BOUNDARY MICROPHONE ADAPTER - A boundary microphone that can reduce the change in its directional property caused by change of a sound collection axis of a microphone unit is provided. The boundary microphone includes a unidirectional microphone unit, a cylindrical unit holding member having a unit accommodating pocket in its peripheral surface to accommodate the unidirectional microphone unit, and a boundary plate to which top face the unit holding member is attached so as to rotate about its axis. When the unidirectional microphone unit is held in the unit holding member, a front acoustic terminal is positioned to face the outside of the peripheral surface of the unit holding member, the sound collection axis intersects the axis of a hollow of the unit holding member, and a rear acoustic terminal communicates with the outside at both side ends of the hollow via the hollow of the unit holding member. | 02-11-2016 |
20160044411 | SIGNAL PROCESSING APPARATUS AND SIGNAL PROCESSING METHOD - A signal processing apparatus acquires an audio signal of channels using a sound acquisition unit, at least a part of which is within a housing of the apparatus, and obtains an audio signal of channels from a microphone provided outside the housing. The apparatus processes an audio signal in accordance with a first propagation characteristic indicating propagation of sound associated with a direction of a sound source, in a case of processing an audio signal acquired by the sound acquisition unit and processes an audio signal in accordance with a second propagation characteristic different from the first propagation characteristic, in a case of processing an audio signal obtained by the microphone. The signal processing apparatus estimates a sound source direction using an audio signal processed by the first processing unit or an audio signal processed by the second processing unit. | 02-11-2016 |
20160057522 | METHOD AND APPARATUS FOR ESTIMATING TALKER DISTANCE - An audio capture device generates two microphone beam patterns with different directivity indices. The audio capture device may determine the position of a user relative to the audio capture device based on sounds detected by the separate microphone beam patterns. Accordingly, the audio capture device allows the determination of the position of the user without the complexity and cost of using a dedicated listening device and/or a camera. In particular, the audio capture device does not need to be immediately proximate to the user (e.g., held near the ear of the user) and may be used to immediately provide other services to the user (e.g., audio/video playback, telephony functions, etc.). The position of the user may include the measured distance between the audio capture device and the user, the proximity of the user relative to another device/object, and/or the orientation of the user relative to the audio capture device. | 02-25-2016 |
20160066083 | METHOD AND APPARATUS FOR MANAGING AUDIO SIGNALS - A method comprising: detect a first acoustic signal by using a microphone array; detecting a first angle associated with a first incident direction of the first acoustic signal; and storing, in a memory, a representation of the first acoustic signal and a representation of the first angle. | 03-03-2016 |
20160066090 | AUDIO DATA PROCESSING METHOD AND ELECTRONIC DEVICE SUPPORTING THE SAME - An electronic device for processing audio data includes: a plurality of microphones configured to receive audio data which has at least one sound section where sounds of a specified frequency band exist within a specified time interval; and an audio data processing module configured to divide the received audio data by each direction of the sound source and collect at least one sound section from the divided audio data of each direction. | 03-03-2016 |
20160066091 | ELECTRONIC DEVICE INCLUDING A MICROPHONE ARRAY - An electronic device comprising: a microphone array including at least three microphones; and at least one processor configured to: identify a kind of an application that is executed; activate one or more of the microphones in the array based on each microphone's respective position within the electronic device and the type of the application; and capture audio using the activated microphones. | 03-03-2016 |
20160086093 | Passive Tracking of Underwater Acoustic Sources with Sparse Innovations - A system and method involve acoustic source localization using passive sonar and capitalizing on the sparse nature of a source location map (SLM). Two types of sparsity are exploited, namely sparsity in the support of the SLMs and sparsity in the innovations across consecutive SLMs. The first type is motivated by the desire to construct SLMs whose non-zero entries corresponded to locations where sources are present. The second type of sparsity is motivated by the observation that few changes occur in the support of consecutive SLMs. Per time instant, an SLM is obtained as the solution of a regularized least-squares problem, where the regularization terms are chosen to encourage the desired sparse structures in each SLM and the innovations. Each SLM may be obtained via a specifically-tailored, computationally-efficient proximal gradient algorithm. | 03-24-2016 |
20160088392 | 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. | 03-24-2016 |
20160094910 | 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. | 03-31-2016 |
20160134969 | LOW NOISE DIFFERENTIAL MICROPHONE ARRAYS - A differential microphone array includes a number (M) of microphone sensors for converting sound to a number of electrical signals, and a processor, operably coupled to the microphone sensors, to specify a target differential order (N) for the differential microphone array, and wherein M>N+1, specify a steering matrix D comprising N+1 steering vectors, calculate a respective one of a plurality of linearly specify a steering matrix D comprising N+1 steering vectors-constrained minimum variance filters based on the steering matrix, apply the respective one of the plurality of linearly-constrained minimum variance filters to a respective one of the electrical signals to calculate a respective frequency response of the electrical signals, wherein the respective frequency response comprises a plurality of components associated with a plurality of subbands, and sum the frequency responses of the electrical signals with respect to each subband to calculate an estimated frequency spectrum of the sound. | 05-12-2016 |
20160148624 | MICROPHONE ARRAY CONTROL SYSTEM - Problem to be Solved | 05-26-2016 |
20160150316 | 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 (( | 05-26-2016 |
20160150319 | METHOD OF MANUFACTURING MICROPHONE, MICROPHONE, AND CONTROL METHOD THEREFOR - Disclosed are a method of manufacturing a microphone, a microphone, and a control method thereof. The method includes forming a sound sensing module on a main substrate including a first sound aperture such that the sound sensing module is connected to the first sound aperture. The method further include forming a cover for receiving the sound sensing module formed therein with a second sound aperture corresponding to the first sound aperture on the main substrate. The method also includes forming a sound delay filter at a receiving space of the cover to be connected to the second sound aperture. The method also includes forming a semiconductor chip electrically connected to the sound sensing module at the receiving space, to selectively operate the sound delay filter according to a signal output from the sound sensing module. | 05-26-2016 |
20160157013 | LISTEN TO PEOPLE YOU RECOGNIZE | 06-02-2016 |
20160163329 | METHOD AND APPARATUS FOR MANAGING AUDIO SIGNALS - A method comprising: detect a first acoustic signal by using a microphone array; detecting a first angle associated with a first incident direction of the first acoustic signal; and storing, in a memory, a representation of the first acoustic signal and a representation of the first angle. | 06-09-2016 |
20160165338 | 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. | 06-09-2016 |
20160165339 | MICROPHONE ARRAY AND AUDIO SOURCE TRACKING SYSTEM - A microphone array of three or more microphones may be mounted on a housing or substrate configured to be mounted on or worn by a person. The microphone array may be positioned so that the far field sensing range of the microphone array is unobstructed by the housing or wearer. An accelerometer may be provided and mounted in a location which is fixed with respect to the microphones of the microphone array. 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. Because the microphone array will move with a user, an accelerometer 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/microphone array. | 06-09-2016 |
20160165340 | 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. | 06-09-2016 |
20160165341 | 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. | 06-09-2016 |
20160165342 | 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. | 06-09-2016 |
20160173976 | HANDHELD MOBILE RECORDING DEVICE WITH MICROPHONE CHARACTERISTIC SELECTION MEANS | 06-16-2016 |
20160173978 | Audio Signal Processing Method and Apparatus and Differential Beamforming Method and Apparatus | 06-16-2016 |
20160182997 | Sound Gathering System | 06-23-2016 |
20160192068 | STEERING VECTOR ESTIMATION FOR MINIMUM VARIANCE DISTORTIONLESS RESPONSE (MVDR) BEAMFORMING CIRCUITS, SYSTEMS, AND METHODS - A method of estimating a steering vector of a sensor array of M sensors according to one embodiment of the present disclosure includes estimating a steering vector of a noise source located at an angle 0 degrees from a look direction of the array using a least squares estimate of the gains of the sensors in the array, defining a steering vector of a desired sound source in the look direction of the array, and estimating the steering vector by performing element-by-element multiplication of the estimated noise vector and the complex conjugate of steering vector of the desired sound source. The sensors may be microphones. | 06-30-2016 |
20160198258 | SOUND PICKUP DEVICE, PROGRAM RECORDED MEDIUM, AND METHOD | 07-07-2016 |
20160205467 | NOISE-REDUCING DIRECTIONAL MICROPHONE ARRAY | 07-14-2016 |
20170238101 | OPTICAL FIBER BASED MICROPHONE ARRAY FOR DETECTING ACOUSTIC EMISSIONS GENERATED BY AN AREA OF INTEREST | 08-17-2017 |
20180024809 | AUDIO SIGNAL PROCESSOR | 01-25-2018 |
20180027325 | MICROPHONE | 01-25-2018 |
20190149913 | ASYMMETRIC MICROPHONE ARRAY FOR SPEAKER SYSTEM | 05-16-2019 |
20190149917 | AUDIO RECORDING SYSTEM AND METHOD | 05-16-2019 |
20190149918 | Sound Signal Collection Method and Apparatus | 05-16-2019 |
20220141581 | Wind Noise Reduction in Parametric Audio - An apparatus including circuitry configured to: obtain at least two audio signals from at least two microphones, wherein the at least two audio signals at least in part includes noise which is substantially incoherent between the at least two audio signals; estimate values associated with the noise within the at least two audio signals; process at least one of the at least two audio signals based on the values associated with the noise; and obtain spatial metadata associated with the at least two audio signals for rendering at least one of the at least two audio signals. | 05-05-2022 |