Abstract: A tracking device for use in obtaining information for controlling an execution of a game program by a processor for enabling an interactive game to be played by a user and related apparatus are disclosed.

Abstract: The invention concerns a method for reproducing spatial impression of existing spaces in multichannel or binaural listening. It consists of following steps/phases: a) Recording of sound or impulse response of a room using multiple microphones, b) Time- and frequency-dependent processing of impulse responses or recorded sound, c) Processing of sound to multichannel loudspeaker setup in order to reproduce spatial properties of sound as they were in recording room, and (alternative to c), d) Processing of impulse response to multichannel loudspeaker setup, and convolution between rendered responses and an arbitrary monophonic sound signal to introduce the spatial properties of the measurement room to the multichannel reproduction of the arbitrary sound signal, and is applied in sound studio technology, audio broadcasting, and in audio reproduction.

Abstract: A method and system for mapping acoustic sources determined from a phased microphone array. A plurality of microphones are arranged in an optimized grid pattern including a plurality of grid locations thereof. A linear configuration of N equations and N unknowns can be formed by accounting for a reciprocal influence of one or more beamforming characteristics thereof at varying grid locations among the plurality of grid locations. A full-rank equation derived from the linear configuration of N equations and N unknowns can then be iteratively determined. A full-rank can be attained by the solution requirement of the positivity constraint equivalent to the physical assumption of statically independent noise sources at each N location.

Abstract: A method and means for a sound-reproduction system. The means include a first opening arrangement for a microphone, and a second opening arrangement for a microphone stand, formed in an essentially planar elastic material.

Abstract: A directional microphone array system generally for hearing aid applications is disclosed. The system may employ a broadside or an endfire array of microphones. In either case, the signals generated by the microphone are added using a plurality of summation points that are connected together via a single signal wire or channel. In the case of the endfire array, all but one of the signals is delayed so that the summation of the signals are in phase. The summation of the signals is then used to generate an output signal for a speaker of a hearing aid or the like.

Abstract: There is provided a microphone unit having a housing, a damping element which is fastened to the housing and an adaptor unit for fastening in a flash shoe of a camera. The damping element is provided between the housing and the adaptor unit.

Abstract: A video camera apparatus provided with image capturing means, which includes: audio communication means for receiving an audio signal from a wireless microphone of a subject through short-range wireless data communication means; and recording means for recording image information of the subject captured by the image capturing means and an audio signal received by the audio communication means.

Abstract: A sound transducer (1), comprising at least one acoustically neutral body (2) with which it is possible to associate two sound conveyance elements (9a, 9b) which are shaped approximately like a stylized funnel so as to each form an auricle (10a, 10b), which protrudes outside the acoustically neutral body (2) and is blended with a duct (19a, 19b) with which a three-pole microphone cartridge (25a, 25b) 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 (29a, 29b) of the microphone cartridges are mutually inverted, so that the cold pole of one of the microphone cartridges and the hot pole (28a, 28b) and the ground (30a, 30b) of the other of the microphone cartridges are or can be connected to a same connector or socket (31a, 31b) which is or can be associated with a suitable amplifying and/or recording and/or processing device.

Abstract: A broadside small array microphone beamforming apparatus comprises first and second omni-directional microphones, a microphone calibration unit, and a directional microphone forming unit. The first and second omni-directional microphones respectively convert voice from a desired near-end talker into first and second signals. The second and first omni-directional microphones and the desired near-end talker are respectively arranged at three points of a triangle. The microphone calibration unit receives the first and second signals and correspondingly outputs first and second calibration signals. The directional microphone forming unit receives the first and second calibration signals to generate a first directional microphone signal with a bidirectional polar pattern. The adaptive channel decoupling unit receives the first calibration signal and the first directional microphone signal to generate a first main channel signal and a first reference channel signal for noise detection.

Abstract: The present invention provides a condenser microphone having a proximity sensor consisting of an infrared light emitting diode and an infrared photodetector, the condenser microphone preventing the occurrence of noise and the malfunctioning of the infrared photodetector when the infrared light emitting diode is lighted using an AC component. Specifically, the present invention provides a condenser microphone operated using a predetermined DC power source; an infrared light emitting diode 20 is lighted at a predetermined frequency by an oscillation circuit 22 using an AC component, and a microphone output is turned on and off by an output signal from the infrared photodetector 21 which is tuned to the predetermined frequency. The condenser microphone further includes two DC-DC converters 23 and 24 connected in parallel with the DC power source. One 23 of the DC-DC converters 23 supplies power to the infrared light emitting diode 20 and the oscillation circuit 22.

Abstract: A microphone package includes a carrier, a cap, a first silicon-based microphone, and a first integrated circuit chip. The carrier has a first storage space. The cap has a planar part contacting the carrier and a plurality of flanges extending from the planar part and surrounding the carrier. The first silicon-based microphone is disposed in the first storage space and covered by the cap. The first integrated circuit chip is disposed in the first storage space, electrically connected to the first silicon-based microphone, and covered by the cap.

Abstract: A hands-free loudspeaker system which is capable of achieving high-quality voice amplification without requiring a human speaker to move to a microphone or a microphone to be moved to a human speaker. A microphone whose input level has continued to be above a threshold value for not shorter than a predetermined time period is detected, based on input signals from dispersedly arranged microphones. An input signal from the microphone is selected and outputted to each loudspeaker at an output level or with a delay time, according to a location of the loudspeaker. A preset lowest threshold level is initially set to the threshold value, and an input level of the microphone higher than the threshold value is newly set to the same, while when the input level is lower than the threshold value, a lower value is set to the same in a step-by-step manner.

Abstract: A microphone rubber apparatus mounted on an SMD microphone prepared on a printed circuit board to protect the SMD microphone from an external physical impact and the like in a portable communication device such as a portable terminal and a Personal Digital Assistant (PDA). The microphone rubber apparatus includes a microphone holder portion formed to engage with and wrap up a microphone, a connection portion protruding to one side of the microphone holder portion and delivering a transmission sound to the microphone through an inside thereof, and at least one shock absorbing portion deformably formed around the connection portion to ease an impact caused by an external force. Thus, the shock absorbing portion having a wrinkled shape can be deformed no matter in which direction an impact is applied to the portable communication device, thereby easing the impact and preventing the microphone rubber from being detached from the microphone due to the external impact or during a distribution test.

Abstract: A mounting device is adapted for mounting a sound pickup device on a surface of a sound source, and includes a base member and a fastening unit. The base member is adapted to be connected to the sound pickup device. The fastening unit is adapted to be provided between the base member and the surface of the sound source for positioning the base member on the surface of the sound source.

Abstract: A microphone bar for obtaining audio input for a video conferencing system includes a master module and one or more slave modules. The master module houses a first pair of unidirectional microphones for obtaining audio from opposite sides of the bar, and each of the slave modules houses a second pair of unidirectional microphones for obtaining audio from opposite sides of the bar. Extensions connect the modules together, and at least one bus conducts audio signals through the extension from the microphones to the video conferencing system. The microphones are operable according to a broadside mode of stereo operation when a central axis of the bar is substantially orthogonal to a view angle of the conferencing system. Furthermore, the microphones are operable according to an end-fire mode of stereo operation when the central axis of the bar is substantially parallel to the view angle of the conferencing system.

Abstract: [Problems] To convert a signal of non-audible murmur obtained through an in-vivo conduction microphone into a signal of a speech that is recognizable for (hardly misrecognized by) a receiving person with maximum accuracy.

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: A clip-on electronic recording device with an automatic activation mechanism is disclosed. When a coupling member coupled to the recording device is clipped onto a clothing accessory of a user, the coupling member activates the recording device automatically. In one embodiment, the recording device has a magnet switch that is activated by a magnet in the coupling member when it is clipped. In another embodiment, a mechanical switch is closed when the coupling member physically pushes into the electronic recording module.

Abstract: The invention concerns a method for detecting and minimizing the harmful influence of ultra sound in a listening device having two or more microphones. The energy contents in the microphone signals is determined, and time related changes in energy contents of the microphone signals is analysed to determine whether any of the microphones is subjected to an ultrasound sound-field, and the signal from the microphone with the lowest content of ultrasound is routed through a signal processing device to an output unit. The invention further concerns a listening device having means for choosing the microphone channel with the lowest level of ultrasound.

Abstract: The outer face of a case which accommodates a whole microphone is coated by a material having a heat conductivity which is lower than that of a metal, and a material transforming temperature which is higher than the charge dissipating temperature of a dielectric layer for forming an internal electret, and which is higher than 260° C., so that the internal temperature rise can be mitigated by the thermal resistance and the thermal capacity of the whole interior. When a microphone is attached to an application equipment, particularly, the microphone may be passed through a reflow solder bath for a short time period. A heat-resistant structure which can prevent the function from being impaired by a high temperature in the period is provided.

Abstract: The small array microphone apparatus comprises first and second omni-directional microphones, a microphone calibration unit and a directional microphone forming unit. The first and second omni-directional microphones respectively convert sound from a desired near-end talker into first and second signals. The second and first omni-directional microphones and the desired near-end talker are arranged in a line. The microphone calibration unit receives the first and second signals, calibrates on gain, and correspondingly outputs first and second calibration signals. The directional microphone forming unit receives the first and second calibration signals to output a first directional microphone signal with a predefined directivity according to a control signal and a second directional microphone signal with a fixed directivity for noise detection. Determination of the control signal is based on whether environmental noise power generated by an environmental detection unit, exceeds a predefined threshold.

Abstract: Embodiments of the present invention provide an audio-based position tracking system. The position tracking systems comprises one or more speakers, an array of microphones and a computing device. The speaker is located at a fixed position and transmits an audio signal. The microphone array is mounted upon a moving object and receives the audio signal. The computing device determines a position of the moving object as a function of the delay of the audio signal received by each microphone in the array.

Abstract: A method constructs a location model for locating and tracking sources of acoustic signals. Acoustic training signals are acquired from an acoustic training source at an unknown location in an environment with an array of microphones placed at unknown positions in the environment. From each acoustic training signal, relative acoustic features are extracted to construct a location model that is trained with the relative acoustic features.

Abstract: The invention is based on the idea of providing a detection device which is capable of distinguishing whether a microphone or audio signal processing arrangement connected thereto operates on an analog or digital basis. An analog or digital connection is selected in accordance with that determination procedure. Consequently there is provided a detection device which has a sensing device, a determining device and a selecting device. The sensing device senses an external signal fed to the detection device and forwards that signal to the determining device. The determining device determines whether the sensed external signal was transmitted by an analog or digital signal processing apparatus and transmits a corresponding determination signal to the selecting device. The selecting device then selects either an analog or a digital connection, in dependence on the determination signal.

Abstract: A microphone unit includes: a partition member, including a vibration film configured to be vibrated by sound; a housing, defining an inner space, and formed with a first opening and a second opening, the inner space being divided into a first space and a second space by the partition member, the first space adapted to be communicated with an outer space via the first opening, and the second space adapted to be communicated with the outer space via the second opening; and a shutter, configured to close one of the first and second openings; and an amplifier, configured to: amplify an electric signal at a first gain when the shutter closes the one of the first and second openings; and amplify the electric signal at a second gain that is larger than the first gain when first and second openings are opened.

Abstract: A digital sound processing system suitable for use in a vehicle comprises an interior rearview mirror assembly that includes at least a first microphone and a second microphone. The first microphone generates a first signal indicative of a vocal input with vehicle cabin noise included therewith and the second microphone generates a second signal indicative of a vocal input with vehicle cabin noise included therewith. A digital signal processor is provided that processes the first signal and the second signal and that provides an output signal with reduced vehicle cabin noise. The digital signal processor preferably comprises a microprocessor executing at least one million instructions per second.

Abstract: A system for normalizing and calibrating the microphones of a sound-intensity probe or a composite of such probes, with respect to a stable comparison microphone with known acoustical characteristics. Normalizing and calibrating are performed using an apparatus 57 consisting of a tube with a loudspeaker inserted in one end and a fixture for holding the microphones of the probe together with the comparison microphone in the other end. The comparison microphone has known acoustical characteristics supplied by the manufacturer. Two banks of quarter-wave resonators 83 and 84 are attached to the side of the tube to absorb standing waves. The sound-intensity probe can be either a two-microphone probe used for measuring a single component of the sound-intensity vector or a probe with four microphones in the regular tetrahedral arrangement used for measuring the full sound-intensity vector.

Abstract: In the method according to the invention a signal processing unit receives signals from at least two microphones worn on the user's head, which are processed so as to distinguish as well as possible between the sound from the user's mouth and sounds originating from other sources. The distinction is based on the specific characteristics of the sound field produced by own voice, e.g. near-field effects (proximity, reactive intensity) or the symmetry of the mouth with respect to the user's head.

Abstract: Methods, microphones, and processors are provided for processing ambient sound waves. Ultrasound waves are combined with the sound waves to create heterodyned sound waves. The heterodyned sound waves are detected and, in response, a sound detection signal containing information relating to the heterodyned sound waves is generated. A heterodyned audio signal representing the heterodyned sound waves is generated at least partially based on the sound detection signal, and then an ambient sound signal representing the ambient sounds is derived from the heterodyned audio signal.

Abstract: An electronic device can be operated to detect noise, such as wind noise. A microphone signal is generated by a microphone. Autocorrelation coefficients are determined based on the microphone signal. Gradient values are determined from the autocorrelation coefficients. The presence of a noise component in the microphone signal is determined based on the gradient values.

Abstract: The beginning detection, accommodation and frequency bias properties of the human hearing mechanism have been modeled to create systems that can detect directional transients (“sound events”) in a sound field and localize them. These systems break down a sound field into sound events and non-sound events and separately localize the sound events and non-sound events. Sound events are generally identified according to the frequency bias and beginning detection properties. Once detected, the sound events are generally localized according to differential steering angles (steering angles to which the steady-state signals have been accommodated) or ordinary steering angles, both of which reflect the direction of a sound event indicated during the rise-time of the sound event. When no sound events are detected, non-sound events are localized according to a steering angle that does not reflect rapid motion.

Abstract: A system is provided for receiving audio signals from a plurality of microphones and transferring the audio signals via a common composite signal channel to a receiving unit, such as a mixing console. The system has at least two satellite units, each having a microphone signal input, a composite signal channel input and summing means for summing a microphone signal and a composite signal. The system further has a master unit with a composite signal channel input, and signal converting means for converting said composite signal into a master signal, which is provided to the receiving unit via a master signal output. In the system, each satellite unit is connected to the common composite signal channel, such that the microphone signal received at the respective satellite unit, is added to the composite signal, which is fed to the master unit.

Abstract: In a pen-based computing system, a pair of microphones captures audio concurrently with a smart pen device capturing handwriting gestures. The audio captured by the pair of microphones is processed to produce an enhanced audio recording. Noise cancellation techniques may be applied to reduce noise generated by the smart pen device interacting with the writing surface to produce a higher quality audio recording. In addition, beam steering techniques may be applied to adjust the gain of portions of the audio signal from different directions and allow the user to focus between multiple audio sources.

Abstract: A camera speakerphone having a microphone array may be used for videoconferencing. Example microphone array designs described herein may be used to perform Sound Source Localization (SSL) and improve audio quality of captured audio. In one example, an omni-directional camera speakerphone includes a base having a speaker and at least one microphone. A neck is coupled to the base which is coupled to a head. The head includes an omni-directional camera and at least one microphone.

Abstract: A directional microphone system is disclosed, which comprises circuitry for low pass filtering a first order signal, and circuitry for high pass filtering a second order signal. The system further comprises circuitry for summing the low pass filtered first order signal and the high pass filtered second order signal. A method of determining whether a plurality of microphones have sufficiently matched frequency response characteristics to be used in a multi-order directional microphone array is also disclosed. For a microphone array having at least three microphones, wherein one of the microphones is disposed between the other of the microphones, a method of determining the arrangement of the microphones in the array is also disclosed.

Abstract: A directional microphone array system generally for hearing aid applications is disclosed. The system may employ a broadside or an endfire array of microphones. In either case, the signals generated by the microphone are added using a plurality of summation points that are connected together via a single signal wire or channel. In the case of the endfire array, all but one of the signals is delayed so that the summation of the signals are in phase. The summation of the signals is then used to generate an output signal for a speaker of a hearing aid or the like.

Abstract: In an audio processing arrangement (2) input signals from a plurality of input sources (4,6) are weighted by weighting factors x and y using weighting elements (10, 12). The weighted input signals are combined to a combined signal by an adder (18). The output signal of the adder (18) constitutes the output of the audio processing arrangement. In order to pronounce the signal with the strongest signal, the weighting coefficients (x,y) are controlled to maximize the output signal of the adder (18) under the constraint that the sum of the squares of the weighting coefficients is equal to a constant.

Abstract: A microphone includes one or more transducers positioned in a housing. Circuitry is coupled to the transducer for outputting an electrical signal such that the microphone has a main lobe directed forwardly and attenuates signals originating from the sides. The microphone transducers can advantageously include multiple transducers, which with the circuit produce a desired sensitivity pattern. The microphone can be employed in a vehicle accessory.

Abstract: Spatial noise suppression for audio signals involves generating a ratio of powers of difference and sum signals of audio signals from two microphones and then performing noise suppression processing, e.g., on the sum signal where the suppression is limited based on the power ratio. In certain embodiments, at least one of the signal powers is filtered (e.g., the sum signal power is equalized) prior to generating the power ratio. In a subband implementation, sum and difference signal powers and corresponding the power ratio are generated for different audio signal subbands, and the noise suppression processing is performed independently for each different subband based on the corresponding subband power ratio, where the amount of suppression is derived independently for each subband from the corresponding subband power ratio. In an adaptive filtering implementation, at least one of the audio signals can be adaptively filtered to allow for array self-calibration and modal-angle variability.

Abstract: Methods for generating a directional sound environment include providing a headgear unit having a plurality of microphones thereon. A sound signal is detected from the plurality of microphones. A transfer function is applied to the sound signal to provide a transformed sound signal, and the transformed sound signal provides an approximation of free field hearing sound at a subject's ear inside the headgear unit.

Abstract: A method and apparatus are provided for matching an output of an auxiliary signal transducer with a reference signal transducer where the auxiliary signal transducer and reference signal transducer receive audio signals from a common signal source along different respective signal paths. The method includes the steps of determining a signal amplitude output value provided by the auxiliary and by the reference transducers within each of a plurality of different frequency ranges in response to the audio signal received along the respective signal paths and adjusting the signal amplitude output value of the auxiliary transducer within at least some of the plurality of different frequency ranges based upon the respective signal amplitude output value of the reference transducer.

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

Abstract: A microphone module includes a microphone and a boot. The microphone has a front surface and a rear surface. The boot includes an internal storage space with the microphone disposed therein, an acoustic passage connected to the internal storage space, and an opening also connected to the internal storage space. The acoustic passage runs substantially parallel to the front surface of the microphone, and the opening exposes the rear surface of the microphone.

Abstract: A method (10) and system (200) for user interface enhancement using speaker directionality can include a speaker phone having a microphone array (35, 36, and 37) on a communication device (32) at a first location and a processor (202) coupled to the microphone array. The processor can be programmed to associate (12) a speaker direction with a given speaker using the microphone array and identify (14) the given speaker and provide an indication of the given speaker at a communication device (42) at a second location in communication with the communication device at the first location. The processor can be further programmed to map (18) or assign sectors to a number of speakers using the microphone array in the first location based on speaker directionality of each of the number of speakers.

Abstract: The present invention increases the aperture size of a microphone array by introducing a diffracting structure into the interior of a microphone array. The diffracting structure within the array modifies both the amplitude and phase of the acoustic signal reaching the microphones. The diffracting structure increases acoustic shadowing along with the signal's travel time around the structure. The diffracting structure in the array effectively increases the aperture size of the array and thereby increases the directivity of the array. Constructing the surface of the diffracting structure such that surface waves can form over the surface further increases the travel time and modifies the amplitude of the acoustical signal thereby allowing a larger effective aperture for the array.

Abstract: The invention relates to a sound recording device for a public address system. Sound is emitted from a sound source, picked up by at least two sound sensors and converted into electrical signals. The sound sensors are located at a distance from a reference position which corresponds to an ideal set position. Directional vectors located between said reference position and the sound sensors indicate different directions. The sound sensors are electrically or acoustically connected to a common signal amplitude add device.

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: In a method of separating acoustic signals from a plurality of sound sources comprising the following steps: disposing two microphones (MIK1, MIK2) at a predefined distance (d) from one another; picking up the acoustic signals with both microphones (MIK1, MIK2) and generating associated microphone signals (m1, m2); and separating the acoustic signal of one of the sound sources (SI) from the acoustic signals of the other sound sources (S2) on the basis of the microphone output signals (m1, m2), the proposed separation step comprises the following steps: applying a Fourier transform to the microphone output signals in order to determine their frequency spectra (M1, M2); determining the phase difference between the two microphone output signals (m1, m2) for every frequency component of their frequency spectra (M1 , M2); determining the angle of incidence of every acoustic signal allocated to a frequency of the frequency spectra (M1, M2) on the basis of the relative phase angle and the frequency; generating a sig

Abstract: A microphone module is externally connected to an electronic device which includes a first sound input jack. The microphone module includes a microphone array, a digital signal processor, and a first connecting part. The microphone array receives sound and generates a first electrical signal. The digital signal processor receives the first electrical signal, performs beam forming and noise suppression, and obtains a second electrical signal. The first connecting part is inserted into the first sound input jack, transmitting the second electrical signal to the electronic device.

Abstract: A sound collecting device is provided which is designed to minimize adverse effects on an output caused by exposure of an electroacoustic transducer to the air. The device includes an electroacoustic transducer and a vibrating circuit. The transducer is exposed to the air and responsive to input of a sound wave to produce a corresponding acoustic signal. The vibrating circuit vibrates the transducer to shake foreign substances such as dust or drops of water from the transducer. In a modified form, an electromagnetic sensor is provided which measures an electromagnetic noise transmitted to the transducer and which removes the electromagnetic noise from an output of the transducer to produce a noiseless acoustic signal.