Abstract: A method for inspection includes capturing an optical image of one or more features on a surface of a sample and irradiating an area of the sample containing at least one of the features with an X-ray beam. An intensity of X-ray fluorescence emitted from the sample in response to the irradiating X-ray beam is measured. The optical image is processed so as to extract geometrical parameters of the at least one of the features and to compute a correction factor responsively to the geometrical parameters. The correction factor is applied to the measured intensity in order to derive a property of the at least one of the features.

Abstract: An X-ray fluorescence analyzer includes a sample stage having an opening at an X-ray irradiation position, an X-ray source which irradiates a sample placed on the opening with a primary X-ray from below, a detector which detects an X-ray fluorescence generated from the sample, a transparent drop prevention plate supported to be advanced and retracted immediately below the opening, a drive mechanism which advances and retracts the drop prevention plate, an observation camera which observes the drop prevention plate positioned immediately below the opening, and an operation unit which processes an image of the drop prevention plate which is captured by the observation camera. The operation unit detects a foreign matter on the drop prevention plate based on an image difference between images before and after the drive mechanism moves or vibrates the drop prevention plate within an observation range of the observation camera.

Abstract: An interferometer includes a diffraction grating that forms a first pattern by diffracting X-rays; a shield grating that forms a second pattern by blocking one or more of the X-rays forming the first pattern; a detector that detects information on the second pattern by detecting X-rays from the shield grating; and a scanning unit that shifts relative positions of an object and a measurable range. In the interferometer, the detector acquires a first detection result by performing a detection while the measurable range and the object take first relative positions and acquires a second detection result by performing a detection while the measurable range and the object take second relative positions. In the interferometer, the scanning unit shifts the relative positions of the measurable range and the object so that a pattern of the first detection result and a pattern of the second detection result pattern have continuity.

Abstract: The present invention proposes a high speed radiographic system for use with megavolt linear-accelerator pulsed x-ray sources to produce video images of large-area fields. A linear accelerator is positioned above a field of view. X-ray photons are directed through an object of interest traveling and/or colliding within the field of view. A large area scintillator system, either truly continuous or in large continuous adjacent pieces, converts the x-ray photons that pass through the object into visible light, and an arrangement of cameras, focused at that plane, where each camera sees a sub-area of the entire scintillator, and these sub-areas overlap somewhat to cover the entire scintillator. The resulting images generated in each camera are synchronized to produce one contiguous, synchronized stream of images.

Abstract: A radiation photographing apparatus 1 includes a radiation irradiating unit 11, a first grating 31 which has a periodic structure which is formed by arranging a plurality of linear bodies 31b and forms a radiation image including a periodic intensity distribution by the passing radiation; a detecting unit 14 which includes a radiation image detector 30 and acquire radiation image data by detecting the radiation image which is irradiated from the radiation irradiating unit to transmit the first grating and a subject and subjected to modulation for the periodic intensity distribution; and a guide unit 70 which guides the arrangement of the subject in relation to an arrangement direction of a group of the linear bodies of the first grating.

Abstract: A method for characterizing a material, comprising: arranging a piece of the material near a source of ionizing photons and a detector; irradiating the piece with photons and acquiring, via the detector, two energy spectra of a photon flux that has been diffused into the material at various depths, the ratio of the photon paths in the material before and after diffusion remaining constant; determining a combined attenuation function with the spectra and the paths; selecting a plurality of energy ranges from said function; calculating, in each range, a quantity that is representative of the function; and estimating, from at least two of said quantities, a physical characteristic of the material by comparison with the same quantities obtained from known materials.

Abstract: Apparatus for wirelessly controlling a guided imaging system based upon the relative motion of the user. The system includes a power supply, a memory, an x-ray source, an image intensifier and a wireless transceiver coupled to the image intensifier. A separate wireless input device comprising a wireless transmitter for communicating with the wireless transceiver of the imaging system may comprise one or more sensors capable of detecting force and directional movement. This detection of movement may then be transmitted to the imaging system and translated into position signals that may direct movement and position of the image intensifier (I-I) as part of the imaging system.

Abstract: A method of processing telephone voice output is applied in an electronic device with telephone function. When any one of the two communicators is a hearing-impaired user, at least one of the electronic devices will obtain the corresponding voice adjustment parameters according to the receiver identification of the hearing-impaired user. Therefore, the voice adjusting program is able to process the voice in advance based on the voice adjustment parameters to help the hearing-impaired user hear better.

Abstract: A method and system for using conversational biometrics and speaker identification and/or verification to filter voice streams during mixed mode communication. The method includes receiving an audio stream of a communication between participants. Additionally, the method includes filtering the audio stream of the communication into separate audio streams, one for each of the participants. Each of the separate audio streams contains portions of the communication attributable to a respective participant. Furthermore, the method includes outputting the separate audio streams to a storage system.

Abstract: A data processing device includes: a receiving unit that receives data; a transfer processing unit that specifies a destination of the data received by the receiving unit, and in accordance with an information processing device of the specified destination, determines a data form of the data to be transferred to the information processing device; a data conversion unit that converts the data received by the receiving unit into the data form determined by the transfer processing unit; and a sending unit that sends the data converted by the data conversion unit, via a circuit capable of connecting to the information processing device of the destination specified by the transfer processing unit, to the information processing device.

Abstract: Detailed information about a party can be provided during interparty communication. Data-capable mobile devices query at least a public database to retrieve information about a set of parties engaged in communication. The results of the database query are employed to provide detailed caller identification and detail functionality to telephones or communication devices not utilizing traditional landlines or commercial caller identification service. Storage can be included to include address book function, query caching, and user preferences. An interface can be provided to facilitate display of rich caller identification information. The interface can further update displayed information without user input, or provide user selection control to request more information on a selected portion of displayed information. These features can be applied to determine the information about a party that is either receiving a communication or initiating a communication.

Abstract: The risk that an incoming emergency call is a prank call is presented in real-time to the called party, in the form of, e.g., an index of suspicion. An index of suspicion aggregation server quantifies, qualifies, and aggregates risks, suspicions, and/or threats relevant to a call center to formulate a scale of suspicion. Metrics are acquired (e.g., calling device location information, device configuration information, caller information, etc.) relevant to each communications request made to a call center and uses acquired metrics to rate communications requests based on a known scale of suspicion. Once a suspicion rating is determined for a communications request, the communications request is routed to a relevant call-taker terminal, as is the index of suspicion for that incoming call. Real time presentation of the index of suspicion enables call-takers to make better call-handling decisions based on risk analysis and historical call data associated with a given caller.

Abstract: A method for monitoring and providing access to back to back scheduled conference calls within a teleconferencing system. The method includes receiving a call and a password entry from a caller. The method includes determining the password entry is valid for a scheduled conference call. The method includes determining the scheduled conference call is scheduled back to back with a previous conference call and that the previous conference call is not over. The method then includes determining the password entry is a specific password, wherein the specific password is issued only to callers accessing back to back scheduled conference calls and placing the caller on hold.

Abstract: In accordance with an embodiment, the logical exclusive operation on the random number RAND and the OPc value are performed several times, the logical operation and encryption processing using a random number RAND and logical operation and encryption processing using a random number S are done in any order, which brings an increase of an attack complexity as to whether to calculate the correlation coefficient from the power consumption waveform at any point when analyzing the correlation power, whereby it is possible to defend the CPA attack effectively.

Abstract: A method for generating a one-way function, as well as a circuit arrangement, which implements the one-way function, are set forth.

Abstract: A device and/or computer program uses a method including determining the division remainder of a first value (b) modulo a second value (p?) and executing a first Montgomery multiplication with the first value (b) as one of the factors and the second value (p?) as a module. A correction factor is determined, and a second Montgomery multiplication is executed with the result of the first Montgomery multiplication as one of the factors and the correction factor as the other factor and the second value (p?) as a module. A method for ascertaining prime number candidates includes determining a base value (b) for a sieve, and several sieve iterations are executed, in which respectively one marking value (p?) is ascertained and multiples of the marking value (p?) in the sieve are marked as composite numbers.

Abstract: Provided are methods and systems of using division-free duplexing (DFD) in a cable communication network. Techniques for applying DFD in a cable communication network may enable data to be transmitted and received over a coaxial cable without using division duplexing techniques. For example, the cable communication network may include DFD enabled network nodes and each subscriber to the cable network may be equipped with a DFD system configured to operate in a DFD mode. In some embodiments, oppositely propagating signals may be transmitted over one frequency channel, and DFD techniques may be used to recover originally transmitted signals. Further, in some embodiments, DFD techniques may be used with encryption methods to increase the security of data transmitted in the cable communication network.

Abstract: An inexpensive radio communication apparatus is provided which can safely generate and share a key to be used for encryption/decryption and lower the possibility of loss of the key even if a packet error occurs at the time of reestablishment of a radio link. When a request for position registration that is made to a base unit is not a request made by encrypted communication using a common encryption key, the base unit transmits a message to request generation of an initial encryption key, and when a handset correctly responds to the message, both the base unit and the handset store information on the initial encryption key.

Abstract: Disclosed is a radio frequency identification (RFID) tag comprising: an RFID functional portion configured to enable wireless communication between the RFID tag and an RFID reader; a data processing functional portion with asymmetric cryptographic capability; and a power source configured to power the data processing functional portion.

Abstract: Access techniques using a mobile communication device are described. In implementations, a mobile communication device comprises a processor, hardware configured to implement multi-mode wireless communication in which at least one of the modes involves telephone communication, tamper-resistant hardware implementing a secure element as storing one or more credentials, and memory having instructions stored therein. The instructions are executable by the processor to cause the mobile communication device to perform operations comprising forming a communication having data that was generated using the one or more credentials stored in the secure element, the communication to be transmitted wirelessly using the hardware implementing the multi-mode wireless communication to initiate function of a physical lock or vehicle.

Abstract: Spherical microphone arrays capture a three-dimensional sound field for generating an Ambisonics representation, where the pressure distribution on the surface of the sphere is sampled by the capsules of the array. The impact of the microphones on the captured sound field is removed using the inverse microphone transfer function. The equalisation of the transfer function of the microphone array is a big problem because the reciprocal of the transfer function causes high gains for small values in the transfer function and these small values are affected by transducer noise. The invention minimises that noise by using a Wiener filter processing in the frequency domain, which processing is automatically controlled per wave number by the signal-to-noise ratio of the microphone array.

Abstract: The present invention discloses a microphone detection circuit for detecting whether an external device includes a microphone. The microphone detection circuit comprises: an audio circuit for receiving an analog input signal which is a signal from the external device or a preset signal. The audio circuit comprises: an analog-to-digital conversion unit to generate a digital audio signal according to the analog input signal; a decision unit, coupled to the analog-to-digital conversion unit, to determine whether the digital audio signal satisfies a predetermined threshold to thereby generate an analysis result, wherein if the digital audio signal satisfies the predetermined threshold, the analysis result indicates that the external device includes a microphone; and a control unit, coupled to the decision unit, to control an operation of the audio circuit according to the analysis result.

Abstract: The acoustic output apparatus includes an output converter that converts an input signal into a plurality of converted signals and outputs the converted signals. The acoustic output apparatus includes a plurality of amplifiers that are provided in a one-to-one relationship with the plurality of converted signals and controlled by the output converter and each of which outputs an output signal obtained by amplifying the converted signal input thereto from the output converter to a load connected to an output thereof. The acoustic output apparatus includes a detector that outputs a detection result to the output converter if the detector detects a failure at the output of any of the plurality of amplifiers.

Abstract: Mobile device speaker control may include: monitoring one or more devices coupled (e.g., wired or wirelessly) with a data network, receiving one or more data packets from each of the one or more devices, filtering received data packets by evaluating a received signal strength (e.g., RSSI) of the received packets. The received packets may be ordered in a priority based on a value, and comparing the received signal strength of each of the received packets to a threshold to determine whether the one or more devices are to perform an action; and/or detecting a device within a proximity of a speaker box coupled with a data network, filtering a data packet received from the device to determine a received signal strength associated with the device, comparing the received signal strength to a threshold, and determining whether an action is to be performed based on a result of the comparing.

Abstract: Methods, systems, and apparatuses are described for improved multi-microphone source tracking and noise suppression. In multi-microphone devices and systems, frequency domain acoustic echo cancellation is performed on each microphone input, and microphone levels and sensitivity are normalized. Methods, systems, and apparatuses are also described for improved acoustic scene analysis and source tracking using steered null error transforms, on-line adaptive acoustic scene modeling, and speaker-dependent information. Switched super-directive beamforming reinforces desired audio sources and closed-form blocking matrices suppress desired audio sources based on spatial information derived from microphone pairings. Underlying statistics are tracked and used to updated filters and models. Automatic detection of single-user and multi-user scenarios, and single-channel suppression using spatial information, non-spatial information, and residual echo are also described.

Abstract: An echo cancelling device splits a low-band signal through LPFs having characteristics which do not cause aliasing during downsampling of downsamplers, and splits a high-band signal through HPFs having characteristics which do not cause aliasing during downsampling of downsamplers. The echo canceling device generates a mid-band signal by subtracting the low-band signal and the high-band signal from a pre-split signal by adder-subtractors, and cancels an echo on a band-by-band basis.

Abstract: A system and method for reducing noise caused by two or more rotating devices by taking in input signals with frequencies that are related to the rotation rates of the rotating devices, and causing one or more loudspeakers to produce sounds that are at about the same frequencies as the noise and of substantially opposite phase. There is a noise canceller associated with each rotating device. Each noise canceller includes a harmonic frequency computer that computes a harmonic frequency and provides the harmonic frequency to a harmonic sine wave generator that generates an output sine wave. Each nose canceller also has an adaptive filter that uses a sine wave to create a noise reduction signal that is used to drive one or more transducers with their outputs directed to reduce noise caused by the rotating devices. There is an overlap detector that compares the harmonic frequencies and, based on their proximity, alters the operation of one or more adaptive filters.

Abstract: Provided is an active vibration noise suppression apparatus having good robustness without determining a secondary transfer function while not performing adaptive control. A first control unit generates a control signal by a first adaptive algorithm as a direct adaptive algorithm. A storing unit stores an equivalent value Gh of a secondary transfer function G which is adaptively updated as an adaptive filter by the first control unit. A second control unit uses a second adaptive algorithm which uses the equivalent value Gh of the secondary transfer function G stored in the storing unit as a feedforward coefficient and updates a filter coefficient C2 of the control signal as an adaptive filter based on the feedforward coefficient. The second control unit generates the control signal by the second adaptive algorithm when the first control unit does not generate the control signal.

Abstract: According to one embodiment, a loudspeaker is provided in a case of an earphone. The case closes an external auditory canal extended from a tympanum of a listener. The earphone has an opening toward the external auditory canal. In an apparatus for generating a sound reproduction to the loudspeaker, a storage unit stores a correction filter in which a maximum of a gain at a frequency band lower than or equal to 10 kHz is larger than a maximum of a gain at a frequency band higher than 10 kHz. An acquisition unit acquires a first sound reproduction signal. A correction unit generates a second sound reproduction signal by convoluting the correction filter with the first sound reproduction signal. An output unit outputs the second sound reproduction signal to the loudspeaker.

Abstract: An audio playback system used in a handheld electronic device is provided. The audio playback system comprises an audio interface, a control module, an audio processing module and at least two audio playback modules. The audio interface receives a digital audio stream. The control module generates audio channel mapping information. The audio processing module processes the digital audio stream to convert the digital audio stream to a plurality of analog audio signals and performs a channel switching process according to the audio channel mapping information. The audio playback modules output the analog audio signals after the channel switching process.

Abstract: A radio module may be configured to receive a starter motor relay signal from a first control module indicative of a motor start. The module may in turn mute a sound device associated with the radio module and upon receiving a delayed accessory signal from a second control module indicative of a delayed accessory trigger, un-mute the sound device in response to receiving the delayed accessory signal.

Abstract: A microphone array arrangement structure in a vehicle cabin is provided with multiple microphones arranged in a convex line. The microphones include a first microphone, and the direction of the axis of the first microphone faces the mouth of the driver. The microphone array arrangement structure is provided with covers housing the microphones and having multiple holes. The front surface of the covers form a convexly curved line, and each of the microphones is fixed inside of a cover to correspond to one of the holes.

Abstract: A vehicular active vibrational noise control apparatus includes an amplitude limiter for limiting the amplitude of a canceling signal based on a signal level of an audio signal, and a vehicle speed detector for detecting the vehicle speed of a vehicle, which incorporates therein the vehicular active vibrational noise control apparatus. The amplitude limiter changes an amplitude limitation rule, which represents a relationship of a limiting value for the amplitude of the canceling signal to the signal level, depending on the vehicle speed, and limits the amplitude of the canceling signal based on the limiting value determined according to the amplitude limitation rule.

Abstract: A mixer includes a plurality of input channels, a plurality of input groups, and a plurality of group strips that adjust values of parameters of the input groups. The mixer preliminarily prepares a plurality of group presets having data required for setup of group modules as presets, and each group preset has attribute information including musical genre, number of input channels, and instrument kind. By selecting a musical genre, a user can change the group presets which are being applied at present to the plurality of input groups to other group presets corresponding to another musical genre while keeping the number of input channels and the instrument kinds which are being used at present between before and after the change of genre.

Abstract: Technology for grouping, consolidating, and pairing individual playback devices with network capability (players) to stimulate a multi-channel listening environment is disclosed. Particularly, the embodiments described herein enable two or more playback devices to be paired, such that multi-channel audio is achieved. Such embodiments may be used to produce stereo and multi-channel audio environments for television and movies.

Abstract: An interpolation circuit for interpolating a first and a second microphone signal and for generating an interpolated microphone signal includes a first input (100) for receiving the first microphone signal (am), a second input (101) for receiving the second microphone signal (am+1), an output (102) for outputting the interpolated microphone signal (s), a control input (103) for receiving a control signal (r), and a first circuit branch (104) including first (105) and second (106) inputs coupled to the first (100) and the second (101) input, respectively, of the interpolation circuit, and an output (107) coupled to the output (102) of the interpolation circuit, wherein the first circuit branch is provided with a means (108) for power-specific summing of the signals supplied to the first and second inputs of the first circuit branch and for outputting a power-specific summation signal at the output (107) of the first circuit branch (104).

Abstract: A microelectromechanical sensing structure for a capacitive acoustic transducer, including: a semiconductor substrate; a rigid electrode; and a membrane set between the substrate and the rigid electrode, the membrane having a first surface and a second surface, which are in fluid communication, respectively, with a first chamber and a second chamber, respectively, the first chamber being delimited at least in part by a first wall portion and a second wall portion formed at least in part by the substrate, the second chamber being delimited at least in part by the rigid electrode, the membrane being moreover designed to undergo deformation following upon incidence of pressure waves and facing the rigid electrode so as to form a sensing capacitor having a capacitance that varies as a function of the deformation of the membrane. The structure moreover includes a beam, which is connected to the first and second wall portions and is designed to limit the oscillations of the membrane.

Abstract: Improved speakers are better able to accurately reproduce sound through the use of low-mass transducers.

Abstract: HRTF on a side of the sound source on the acoustic signal. A crosstalk compensation processing unit performs, with respect to the first binaural signal and the second binaural signal, a crosstalk compensation for canceling out an acoustic transfer characteristic and a crosstalk. The present technology can be applied to, for example, an AV amplifier.

Abstract: A hearing instrument includes a housing and an exchangeable receiver. The hearing instrument further includes an opening in the housing for receiving the exchangeable receiver therein, and a coupler for actuating/de-actuating a physical coupling between the housing and the exchangeable receiver and for enabling the housing to hold the receiver.

Abstract: A method, including determining a change in an actuator impedance based on a change in an electrical property of a system of which the actuator is apart, and determining one or more system characteristics based on the change in the actuator impedance.

Abstract: Hearing systems for both hearing impaired and normal hearing subjects comprise an input transducer and a separate output transducer. The input transducer will include a light source for generating a light signal in response to either ambient sound or an external electronic sound signal. The output transducer will comprise a light-responsive transducer component which is adapted to receive light from the input transducer. The output transducer component will vibrate in response to the light input and produce vibrations in a component of a subject's hearing transduction pathway, such as the tympanic membrane, a bone in the ossicular chain, or directly on the cochlea, in order to produce neural signals representative of the original sound.

Abstract: This disclosure relates generally to an apparatus for retaining an earpiece in the ear during physical movement and exercise. In various instances, the apparatus may include a thinned region a posterior arch and ribs to allow deformation of the earmold and facilitate securing of the earmold in the concha bowl of the ear.

Abstract: A simple hearing enhancement device that takes the normally adequately loud sound levels and optimizes selective frequency gain of the patient's ear passage to improve speech comprehension.

Abstract: Techniques for managing a network of speaker lights and wearable devices using intelligent connection managers are described. Disclosed are techniques for receiving data representing a distance between a wearable device and a speaker light, the speaker light associated with an identifier, and generating an audio control signal and a light control signal as a function of the distance. The audio control signal may include data representing an audio parameter and data representing the identifier, and the light control signal may include data representing a light parameter and data representing the identifier. Presentation of an audio signal using the audio parameter and a light using the light parameter may be caused at the speaker light. The audio signal and the light may be substantially directed towards the wearable device or a user.

Abstract: The present invention comprises a method and apparatus for installing a tile-mounted ceiling speaker that combines the ease of installation of a lay-in speaker system with the visual appearance of a tile-mounted speaker system. In one or more embodiments, the apparatus of the present invention includes a support frame and back box assembly configured for installation on top of a ceiling tile and a loudspeaker cartridge configured to be mounted to the support frame through an appropriately-sized hole in the ceiling tile.

Abstract: 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.

Abstract: A closed-back headphone includes a housing (111b) and an ear pad (111a). The housing (111b) includes a speaker unit. The ear pad (111a) is mounted on the housing (111b) and is a ring-shaped cushion surrounding an ear of a user (401). The ear pad (111a) includes an upper part (P1) for covering an upper portion of the ear, and a lower part (P2). The upper part (P1) is softer than the lower part (P2).

Abstract: An earphone including a housing, a speaker, a tuning mechanism and an acoustic damper is provided. The housing has a sound-output opening. The speaker is disposed at the sound-output opening. The speaker and the housing together define a compartment. The speaker has an inner chamber and a through hole for communicating the inner chamber and the compartment. The tuning mechanism is movably disposed at the housing and extended into the compartment. The acoustic damper is disposed between the tuning mechanism and the speaker, and located at the through hole. The tuning mechanism is configured to move to compress or release the acoustic damper for adjusting the air permeability of the acoustic damper.

Abstract: The invention disclosed is an improved audio output module for use with an artificial voice generation device, having a housing separated into a sound system chamber, an interface chamber, and a power source chamber. The interface and power source chambers may be combined. The sound chamber is isolated from external air by the housing, the cover plate, and a separating wall, which separates it from other chambers of the module. Volumetric parameters based on speaker characteristics and design requirements can thus be implemented independent from the choice of interface type. The module is configurable to be mounted to an external structure or to a speech generating system. It may likewise be detachable from a quick release cradle and receive wireless audio signals from the speech generating system.