Abstract: A sound input device includes a differential microphone, configured to receive sound including noise, and generate a first signal in accordance with the sound; a detector, configured to detect the noise, and generate a second signal in accordance with the detected noise; and a controller, configured to control at least one of suppression of high-frequency components of the first signal and changing of a frequency band to be suppressed of the first signal based on the second signal.

Abstract: A voice signal transmitting/receiving apparatus includes: a device body; a speaker array arranged in the device body and including a plurality of arrayed speaker units; and a microphone array arranged in the device body and including a plurality of arrayed microphones. By thus integrating the speaker array and the microphone array, it is possible to improve the operability of a user, to acquaint the user relatively easily with the error in the set position and to make the device compact.

Abstract: The invention provides a computer-implemented method for determining a time delay for time delay compensation of a microphone signal from a microphone array in a beamformer arrangement. For a given time, an instantaneous estimate of a position of a wanted sound source and/or of a direction of arrival of a signal originating from the wanted sound source is determined. The computer system then determines whether the instantaneous estimate deviates from a preset estimate of a position of the wanted sound source and/or of a direction of arrival of a signal originating from the wanted sound source according to a predetermined criterion. The predetermined criterion comprises a check whether the instantaneous estimate deviates from the preset estimate by at least a predetermined deviation threshold.

Abstract: 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.

Abstract: A portable sound recorder has a body case, a left microphone and a right microphone held rotatably by the body case. An inwardly arcuate microphone receiving cavity is formed at each upper corner of the body case. The microphone receiving cavity is configured to elevate a rear side end of its arcuate bottom surface, and surrounds the under side and a part of the rear side of the microphone. Formed on the bottom surface of the microphone receiving cavity is a hollow portion that creates a gap between the microphone receiving cavity and the microphone, so as to prevent the reflected sound off the microphone receiving cavity from reaching the microphone. The gap is maintained by the hollow portion even when the microphone is rotated.

Abstract: Sound waves having a proper phase difference are received by microphones fixed in a mesh-formed casing, while other sound waves pass through the casing, and reach a front surface of a diffuse reflection member. The randomly uneven front surface of the diffuse reflection member diffusely reflects the sound waves, thereby preventing the reflected sound waves from reaching the microphones at the proper phase difference. Any reflected sound waves that do reach the microphones are received at a phase difference that is different from the proper phase difference and are determined to be noise by a sound-source determining circuit, thereby enabling a sound receiver to receive only sound waves having the proper phase difference, and hence, improving directivity thereof.

Abstract: A directional sensor array system generally for remote audio collection applications that is modular, scalable, and robust with the modules assembled in layers. The invention can alternatively employ sensors other than microphones, such as ultrasonic transducers and accelerometers. In the preferred embodiment, the sensors are mounted on tiles, each of which performs its own local beamforming using a low-impedance resistive summation technique. The tiles are constructed in a layered, sandwiched fashion and incorporate integral protection from wind, sand, dust, moisture, radio frequency noise, vibration, ambient acoustic noise, and directional acoustic noise, as well as provide inter-sensor isolation. Multiple tiles can be joined together physically and electrically. When joined, a secondary parallel beamforming is performed on the bus using electrical summation.

Abstract: A sound transducer, comprising at least one acoustically neutral body with which it is possible to associate two sound conveyance elements which are shaped approximately like a stylized funnel so as to each form an auricle, which protrudes outside the acoustically neutral body and is blended with a duct with which a three-pole microphone cartridge is associated, the cartridge being arranged so that its front end, adapted to acquire the sound, is proximate to the inlet of the duct. The two cold poles of the microphone cartridges are mutually inverted, so that the cold pole of one of the microphone cartridges and the hot pole and the ground of the other of the microphone cartridges are or can be connected to a same connector or socket which is or can be associated with an amplifying and/or recording and/or processing device.

Abstract: In an electronic apparatus, microphones are arranged parallel to one side of a display screen so as to be spaced apart from each other on the front surface of a display whose display screen is exposed. Together with the microphone, a microphone is arranged parallel to other side different from one side of the display screen. Of the microphones, the microphones are selected when one side of the display screen serves as a bottom, and the microphones are selected when other side of the display screen serves as the bottom. A voice input process is then executed.

Abstract: 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.

Abstract: 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 (103) which (i) analyze frequencies of the mixed sound including the noise and received by each microphone, and (ii) generates frequency signals; and an extraction unit (105) which, for each source location candidate, (a) adjusts time axes of the frequency signals corresponding to the microphones, so that there is no time difference between when the mixed sound reaches one microphone from the source location candidate and when the mixed sound reaches another microphone from the source location candidate, and (b) determines frequency signals having a difference distance equal to or smaller than a threshold value, from among the frequency signals corresponding to the microphones with the time axis having been adjusted, the difference distance representing a degree of a difference in the frequency signals between the microphones, and (c) extracts the sourc

Abstract: Disclosed herein is a headphone device, including: a sound pickup section configured to pick up an external sound; a directivity setting section configured to generate a directional pickup audio signal, which is an audio signal obtained by picking up the external sound with a desired directional characteristic, based on an audio signal outputted from the sound pickup section; a loudspeaker; an audio signal generation section configured to generate a cancellation-use audio signal for attenuating the directional pickup audio signal based on the directional pickup audio signal; and a driving signal generation section configured to generate a driving signal, which is an audio signal for driving the loudspeaker and includes at least the cancellation-use audio signal.

Abstract: The invention relates to simulation of sound fields in enclosures, for instance for application in listening tests, where test subjects assess the sound quality or other sound perception characteristics of the sound field. According to a specific embodiment, the system comprises a binaural synthesis portion which synthesizes sound for instance from a sound-reproduction equipment based on measured impulse responses of an actual room stored in a data base (31) and a binaural recording portion comprising a data base 32 for storing binaural recordings of other sound signals made in the room. Data from these databases are mixed (41) and reproduced by means of a headphone (39) provided with a head tracker (42) for tracking the movements of the listener's head. The invention furthermore comprises the use of cross-fading functions (36, 37) to enable the dynamic listening conditions, where the movements of the listener's head are taken into account during the simulation process.

Abstract: A remote controlled robot system that includes a robot and a remote control station. The robot includes a binaural microphone system that is coupled to a speaker system of the remote control station. The binaural microphone system may include a pair of microphones located at opposite sides of a robot head. The location of the microphones roughly coincides with the location of ears on a human body. Such microphone location creates a mobile robot that more effectively simulates the tele-presence of an operator of the system. The robot may include two different microphone systems and the ability to switch between systems. For example, the robot may also include a zoom camera system and a directional microphone. The directional microphone may be utilized to capture sound from a direction that corresponds to an object zoomed upon by the camera system.

Abstract: A microphone assembly for desktop communication systems utilizes a directional microphones in a desktop conferencing system without exposing the microphone to unfavorable mechanical or acoustic influence. The microphones is built into the front portion of the base of the system, in a mechanically controlled and robust way. The microphone assembly maximizes microphone sensitivity in the direction of a near end user while simultaneously minimizing microphone sensitivity in the direction of the loudspeaker.

Abstract: 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.

Abstract: A device and method for providing augmented reality enhanced audio. The device is portable and may include a location processing module capable of receiving location information and determining a location of the device in an environment, an orientation processing module capable of determining an orientation of the device at the location, a processor capable of modifying audio content based on the location of the device in the environment and the orientation of the device at the location, and an audio output device capable of outputting the modified audio content. The orientation of the device includes an azimuth and an elevation of the device.

Abstract: A karaoke device is included within an enclosure and having a voice pickup element integrated into the enclosure, the voice pickup element for converting sound waves into an electrical signal. An audio input signal passes into the enclosure and connects with an electronic circuit for amplifying the electrical signal, for controlling the amplitude of the electrical signal and for mixing the electrical signal and the audio input signal into a mixed audio signal which passes out of the enclosure. A video input signal passes into the enclosure and directly connects to a video output signal that passes out of the enclosure.

Abstract: A device for monitoring outside sound through a closed window includes a housing positioned inside the window. The housing contains a filter, a squelch circuit, an amplifier, and a speaker. A microphone is positioned outside the window and connected to the housing. The housing includes a volume control, a sensitivity control, and a mode control for bypassing the squelch circuit.

Abstract: An apparatus and method for communications is disclosed. The apparatus includes a receiver configured to scale an audio signal, and a transducer circuit configured to provide an ambient signal in response to an ambient condition, wherein the receiver is further configured to scale the ambient signal from the transducer circuit and combine the scaled ambient signal with the scaled audio signal, the receiver being further configured to adjust the scaling applied to at least one of the ambient and audio signals.

Abstract: A semiconductor circuit including an input terminal, an impedance converting portion configured to receive an input signal from the input terminal and to output an output signal corresponding to the input signal, an input impedance of the semiconductor circuit being higher than an output impedance of the semiconductor circuit, a detecting portion connected to a node between the input terminal and the impedance converting portion, and configured to detect whether the input signal is higher than a predetermined threshold, and a variable impedance connected to a reference voltage and the node, an impedance of the variable impedance configured to decrease after the input signal is detected as higher than the predetermined threshold.

Abstract: The present invention provides a sound generating method of generating sound signals related to a video signal, which comprises a step of generating independently each of the sound signals matched to a horizontal direction and a vertical direction of a video, and a step of allowing the horizontal and the vertical sound signals that have been generated to be reproduced independently with horizontal sound output means and vertical sound output means, respectively.

Abstract: 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.

Abstract: A method and apparatus for sound reinforcement for musical instruments. The method and apparatus re-direct sound waves via an open-sided chamber against a curved wall panel in such a way that the sound from a musical amplifier is reflected or altered in its course from a direction in which the microphone design has reduced sensitivity to a direction of maximum sensitivity and, typically, altering the direction of the sound from off-axis to on-axis into the microphone capsule.

Abstract: The invention provides an integrated circuit. The integrated circuit receives a first signal from a microphone via a first node. In one embodiment, the integrated circuit comprises a biasing circuit and a buffering circuit. The biasing circuit is coupled between the first node and a second node, drives the microphone with a first voltage source, and filters the first signal to generate a second signal at the second node. In one embodiment, the biasing circuit comprises a first resistor, a first capacitor, and a load element. The first resistor is coupled between the first voltage source and the first node. The first capacitor is coupled between the first node and the second node. The load element is coupled between the second node and a second voltage source. The buffering circuit is coupled between the second node and a third node and buffers the second signal to generate a third signal at the third node.

Abstract: 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.

Abstract: An arbitrarily positioned cluster of three microphones can be used for stereo input of a videoconferencing system. To produce stereo input, right and left weightings for signal inputs from each of the microphones are determined. The right and left weightings correspond to preferred directive patterns for stereo input of the system. The determined right weightings are applied to the signal inputs from each of the microphones, and the weighted inputs are summed to product the right input. The same is done for the left input using the determined left weightings. The three microphones are preferably first-order, cardioid microphone capsules spaced close together in an audio unit, where each faces radially outward at 120-degrees. The orientation of the arbitrarily positioned cluster relative to the system can be determined by directly detecting the orientation or by using stored arrangements.

Abstract: 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 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.

Abstract: A range-sensitive wireless-microphone method includes receiving an audio input, converting the received audio input into digital data, buffering the digital data, and transmitting the buffered digital data. The method also includes determining whether the transmitted buffered data was successfully received, responsive to a determination that the transmitted buffered data was successfully received deleting the transmitted buffered data, and, responsive to a determination that the transmitted buffered data was not successfully received, retaining the transmitted buffered data and repeating the transmitting step. This Abstract is provided to comply with rules requiring an Abstract that allows a searcher or other reader to quickly ascertain subject matter of the technical disclosure. This Abstract is submitted with the understanding that it will not be used to interpret or limit the scope or meaning of the claims.

Abstract: In various embodiments, a speakerphone may comprise multiple (e.g., 16) microphones placed in a circular array around a central speaker. The microphones may be mounted vertically in the speakerphone with their respective diaphragms substantially parallel to the top surface of the speakerphone. The centrally mounted speaker may be coupled to a stiff internal speaker enclosure. The speaker enclosure may be made of a stiff, heavy material (e.g., a dense plastic) to prevent the speaker vibrations from excessively vibrating the speakerphone enclosure (which may affect the microphones). The speaker enclosure may also include a raised rim and ridges to increase its stiffness.

Abstract: Disclosed are an apparatus and a method for beamforming in consideration of characteristics of an actual noise environment. The apparatus includes a microphone array having at least microphone, the microphone array outputting a signal input through the microphone; a coherence function generation unit for calculating coherences for input signals according to each space between microphones, calculating averages of the coherences for the same distance, and filtering the calculated averages of the coherences and outputting the resultant values, when an input signal is input; a spatial filter factor calculation unit for calculating and outputting a spatial filter factor by using the filtered average coherences; and a beamforming execution unit for performing a beamforming for the input signals by using the spatial filter factor, thereby outputting a noise-processed signal.

Abstract: 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.

Abstract: A mouse device includes a mouse base, a mouse casing, a first button, a second button, a microphone module and a controlling unit. The first button and the second button have different structures. By clicking the first button and the second button, a first sound and a second sound having different frequencies are respectively generated. The first sound and the second sound are received by the microphone module. According to the frequency of the first sound or the second sound, a first button command or a second button command is generated.

Abstract: An audio signal processing apparatus includes first, second and third omni-directional microphones each of which receives sound and generates an omni-directional audio signal and which are spaced apart by a predetermined distance, a first adder section that adds audio signals generated by the first, second and third omni-directional microphones and generates an audio signal having an omni-directivity in the whole circumferential direction, a first subtractor section that subtracts audio signals generated by the first and third omni-directional microphones and generates an audio signal having a directivity in the right-left direction, a second adder section that adds audio signals generated by the first and third omni-directional microphones, a second subtractor section that subtracts an audio signal generated by the second omni-directional microphone from the audio signal added by the second adder section and generates an audio signal having a directivity in the front-back direction, and an output section tha

Abstract: Targeted sound detection methods and apparatus are disclosed. A microphone array has two or more microphones M0 . . . MM. Each microphone is coupled to a plurality of filters. The filters are configured to filter input signals corresponding to sounds detected by the microphones thereby generating a filtered output. One or more sets of filter parameters for the plurality of filters are pre-calibrated to determine one or more corresponding pre-calibrated listening zones. Each set of filter parameters is selected to detect portions of the input signals corresponding to sounds originating within a given listening zone and filter out sounds originating outside the given listening zone. A particular pre-calibrated listening zone is selected at a runtime by applying to the plurality of filters a set of filter coefficients corresponding to the particular pre-calibrated listening zone.

Abstract: A micro-electro-mechanical-system microphone package includes a substrate, a micro-electro-mechanical-system microphone chip mounted on the substrate, and a cover attached to the substrate to cover the micro-electro-mechanical-system microphone chip. The cover is provided with a sound inlet through which the micro-electro-mechanical-system microphone receives external sound. The micro-electro-mechanical-system microphone chip includes a conductive base connected to a constant voltage, a shielding layer supported by the conductive base and connected to the constant voltage, a diaphragm disposed between the conductive base and the shielding layer, and a back plate also disposed between the conductive base and the shielding layer.

Abstract: A listening device for presenting at least one audio signal to a user comprises a first ear piece and a second ear piece, each of the first ear piece and the second ear piece including at least one speaker and each being adapted to block substantially all external sounds from entering an ear of a user. The device further comprises a microphone for sensing a first audio signal and a differential amplifier coupled to the microphone for amplifying the first audio signal. The differential amplifier is coupled to a first electrical ground. A switching arrangement coupled to and receiving a second audio signal from an external audio source and the external audio source having a second electrical ground that is electrically distinct from the first electrical ground.

Abstract: A microphone compensation system responds to changes in the characteristics of individual microphones in an array of microphones. The microphone compensation system provides a communication system with consistent performance despite microphone aging, widely varying environmental conditions, and other factors that alter the characteristics of the microphones. Furthermore, lengthy, complex, and costly measurement and analysis phases for determining initial settings for filters in the communication system are eliminated.

Abstract: 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.

Abstract: An in-ear stabilizer with a compression strut securely holds an in-ear communications device within the human ear. The in-ear stabilizer comprises an in-ear device housing which fits into the human ear and has a receiver and speaker. Attached to the in-ear device housing is a compression strut which extends, when in the human ear, across the crus of helix and fits securely under the antihelix in the upper concha of the ear. The housing of the in-ear device fits into the lower concha of the human ear. The compression strut exerts outward pressure when compressed, comfortably holding the in-ear device in place through pressure on the lower and upper concha of the human ear.

Abstract: A method for estimating the location of a sound source. First, a first microphone and a second microphone are moved relative to the sound source. Then the Intra-aural Time Difference (ITD) between the two microphones is measured. The moving and the measuring are repeated until the Intra-aural Time Difference (ITD) is smaller than a predefined threshold.

Abstract: Electronic devices and accessories such as headsets for electronic devices are provided. A microphone may be included in an accessory to capture sound for an associated electronic device. Buttons and other user interfaces may be included in the accessories. An accessory may have an audio plug that connects to a mating audio jack in an electronic device, thereby establishing a wired communications path between the accessory and the electronic device. Path configuration circuitry may be used to selectively configure the path between the electronic device and accessory to support different operational modes. Analog audio lines in the wired path may convey left and right channel analog audio channels. When it is desired to convey power over the wired path, one of the analog audio channel lines may be converted to a power line. Audio functionality may be retained by simultaneously converting a unidirectional line into a bidirectional line using hybrids.

Abstract: A microphone has at least one non-conductive component with at least one side covered by a metal layer which in turn is covered by a protective layer, and the microphone can be a component of a delegate unit of a conference system or a congress system.

Abstract: A handsfree communication system includes microphones, a beamformer, and filters. The microphones are spaced apart and are capable of receiving acoustic signals. The beamformer compensates for propagation delays between the direct and reflected acoustic signals. The filters are configured to a predetermined susceptibility level. The filter process the output of the beamformer to enhance the quality of the received signals.

Abstract: An apparatus having a microphone and a mute circuit (not shown in detail here) operable to be actuated by a rotating door assembly. The apparatus provides a mute circuit suitable to mute the audio microphone when the rotating door assembly is in a closed position and unmute the audio microphone when the rotating door assembly is not in the closed position. Additionally, the mute circuit is impervious to generating audible disturbances in the microphone signal when the rotating assembly door is rotated between open and closed positions. This advantageously allows for the microphone to essentially transition to off when the rotating door assembly is closed and transition to on when the rotating door assembly is open and do so without additional audio disturbance.

Abstract: A microphone apparatus for processing and outputting an output signal of a microphone array including at least nine microphones includes a directivity function processing circuit that converts the output signal of the microphone array into a unidirectional signal and that outputs the unidirectional signal. The directivity function processing circuit expands a directivity function whose variable is an incident angle of an acoustic wave into a Fourier series up to at least third order. The variable in the expanded expression is produced from output signals of the microphones forming the microphone array.

Abstract: 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.

Abstract: An arrangement for mounting a microphone to a surface in a passenger compartment of a vehicle includes, but is not limited to, a housing having an opening. The housing is configured to be mounted to the surface. A directional wideband microphone is mounted within the housing and positioned to partition the housing into a first chamber and a second chamber. The directional wideband microphone has an acoustic axis that extends through the first chamber. The housing and the directional wideband microphone cooperate to direct the acoustic axis in a predetermined direction.

Abstract: A synchronous detection and calibration system is provided for expedient calibration of differential acoustic sensors in a manufacturing and testing environment. By processing a series of sequentially received tones, respective portions of a system using differential acoustic sensors are tuned for optimum individual operation, following which corresponding control data are generated and stored for use in selecting among predetermined calibration vectors which establish and maintain optimum system operation.

Abstract: The invention regards a listening device with two or more microphone units. The listening device has a signal processing device and means for delivering a signal to the user of the device representative of the audio signals picked up by the microphones, whereby the signal processing device comprises means for adding and scaling the signals from at least two microphone units to provide a single added signal in a manner which allows signal parts from different directions to be equally represented in the resulting added signal.