Abstract: A method for geometric calibration of a CT scanner, including, for each row of at least one row of detector cells, establishing a complete geometric description of the CT scanner, including at least one unknown geometric parameter, establishing a description of a forward projection function using the complete geometric description, acquiring actual projection coordinates of a calibration phantom placed in a scanning field of view (SFOV) on a current row of detector cells and corresponding to a plurality of angles, acquiring calculated projection coordinates of the calibration phantom on the current row of detector cells and corresponding to the plurality of angles using the description of the forward projection function, and acquiring a calibrated value for the at least one unknown geometric parameter by evaluating the at least one unknown geometric parameter based on the acquired actual projection coordinates and calculated projection coordinates via a nonlinear least square fitting algorithm.

Abstract: A method and device for transmission line testing are provided. The method includes applying, at a customer premises device, a test signal to a transmission line coupled to the customer premises device. The method further includes measuring, at the customer premises device, a reflection of the test signal. The method also includes determining, at the customer premises device, a characteristic of the transmission line based on the reflection of the test signal. The method includes determining, at the customer premises device, whether the transmission line supports a transmission service based on the characteristic. The method further includes generating an output, at the modem device, the output indicative of whether the transmission line supports the transmission service.

Abstract: Systems and methods are provided to guide an emergency dispatcher in responding to emergency calls involving an active assailant. The systems and methods can include an emergency police dispatch protocol configured to facilitate rapid, uniform, and consistent handling of events involving active assailants. The emergency police dispatch protocol includes one or more dispatch protocols configured for particular incidents or emergency situations, such as to aid the dispatcher in processing calls relating to active assailants. The emergency police dispatch protocol presents a pre-scripted interrogation, including preprogrammed inquiries for a dispatcher to ask the caller. The pre-scripted interrogation of the dispatch protocol facilitates rapid, uniform, and consistent responses to incidents with active assailants. The dispatch protocol may facilitate deployment of responders as quickly as possible to active assailant events and update responders with information as it is gathered.

Abstract: A method and communication device for scheduling a conference call. The method includes receiving, in a first communication device, an invitation message containing at least some conference call scheduling information with respect to a scheduled conference call session; displaying an interface in relation to the invitation message, the interface including an option to send a request for recorded conference call content of the scheduled conference call session; receiving an input selecting the option; and sending a communication to a second communication device including a response to the invitation message and including said request for recorded conference call content. A conference call server is also described for sending recorded conference call content to the communication device.

Abstract: Systems and methods for identifying participants present in a communication session are disclosed. More particularly, the identification of participants using a shared communication endpoint is enabled. Identification can include receiving information from a first participant that identifies a second participant. The identification of the second participant can include receiving a selection of the second participant from a list of expected conference participants presented to the first participant through a communication device associated with the first participant, after the first participant has been registered as a participant in the communication session.

Abstract: A call management device is connected between a standard push button telephone, rotary dial telephone, or other communication device and a public telephone network, and it operates to place (dial) calls on the public telephone network. The call management device maintains a telephone book that can be used to look up a phone number by name. A caller, using their telephone's keypad or rotary dial, can either dial a phone number in the normal fashion or they can dial-by-name from the telephone book.

Abstract: Transformation of binary data files created by a data logger of a telephone system used by a call center into a searchable form for generation of meaningful management reports concerning the operation and performance of a call center, its agents and equipment is accomplished by providing a report server which u, under software control, processes the data in the log file to complete fields of tables in a database. The database is then used to create and deliver management reports both automatically and in response to queries entered by supervisors.

Abstract: A system implements a method of assessing performance of a call center agent that includes identifying a set of the one-time caller calls that the agent has handled. A first issue resolution rate is determined for the calls in the set that were released by the customer, and a second issue resolution rate is determined for the calls in the set that were released by the call center agent. The method also includes determining a difference between the first issue resolution rate and the second issue resolution rate. The determined difference is used to generate a performance assessment for the call center agent.

Abstract: A system and method for monitoring and visualizing contact center performance via a graphical user interface. A processor identifies a plurality of contact center metrics, identifies predicted values for the plurality of contact center metrics, and displays a reference geometric object having a plurality of vertices representing the predicted values for the plurality of contact center metrics. The processor also determines actual values for the plurality of contact center metrics and overlays over the display of the reference geometric object, a display of a current geometric object. The current geometric object has a plurality of vertices representing the actual values for the plurality of contact center metrics. Deviation of the current geometric object from the reference geometric object is an indication of deviation of one or more actual values from the predicted values.

Abstract: The method and system of the present invention include, responsive to a receipt of a customer input, customizing an agent interface menu. The customizing includes selecting a plurality of action items for the agent interface menu, and ordering the plurality or more action items in the agent interface menu. The customizing is performed using the customer input and an item of stored customer information. The method further includes presenting an item of the customer input in an agent interface. The agent interface includes the agent interface menu, a communication window, and a customer attributes display. Responsive to a selection of one of the plurality of action items, wherein the plurality of action items is presented by the agent interface menu, an input field in a form provided in support of an action selected from the agent interface menu is populated with the customer input received from the customer interface.

Abstract: A system that incorporates teachings of the present disclosure may include, for example, a controller to: determine crosstalk coupling characteristics between a plurality of lines of a digital subscriber line system connected to a plurality of modems based on a transition between a full power mode and one or more other modes, and provide the determined crosstalk coupling characteristics to one or more of the plurality of modems for performance of at least one of pre-coding a transmitted signal and processing a received signal along a line of the plurality of lines, where the pre-coding and processing is performed based at least in part on the determined crosstalk coupling characteristics, and where the pre-coding of the transmitted signal and the processing of the received signal reduce effects of fluctuating crosstalk. Other embodiments are disclosed.

Abstract: A reduced-memory vectored DSL system includes methods and apparatus for reducing the bandwidth and memory storage demands on a vectored DSL system in which FEXT data is transmitted and stored. When test signal data, such as training and/or tracking data, is sent to determine FEXT characteristics of the DSL system, error signals are available for all or substantially all of the upstream and/or downstream frequency band DSL tones used in the system. Dividing a frequency band into sub-bands, only a subset of tones in each sub-band is used for deriving FEXT data. For tones in the sub-band subsets, full-precision FEXT data values can be derived. For other tones, approximations of the FEXT data can be derived. Memory is reduced in both the transmission of such FEXT data (between upstream and downstream ends of the DSL system) as well as within an upstream-end device such as a DSLAM that performs vectoring using a separate or internal vectoring processing apparatus.

Abstract: A data processing system includes a module for generating and distributing random masks to a number of cryptographic accelerators while providing for fewer total interconnects among the components generating the random masks. The module segments the tasks associated with generating random masks across a number of modules and blocks such that routing and timing problems can be minimized and layout can be optimized. A method for generating and distributing random masks to a number of cryptographic accelerators is also provided. The random masks are utilized by cryptographic accelerators to protect secret keys, and data associated with those keys, from discovery by unauthorized users.

Abstract: A modified Chinese State Encryption Management Bureau's SM2 Elliptic Curve Signature Algorithm that offers partial message recovery and lowers the signature size for a given cryptographic strength. The modified SM2 Elliptic Curve Signature Algorithm includes a signature and verification algorithm that modifies a signature generation primitive to compute a key derived from the ephemeral signing key, and a multiple of the signer's public key.

Abstract: Techniques for injecting encryption keys into a meter as a part of a manufacturing process are discussed. Since various encryption keys injected into meters may be specific to each individual meter, a utility company customer may require a copy of the injected encryption keys associated with each individual meter. The techniques may include providing a copy of keys injected into each meter to a utility company customer. In some instances, the meter manufacturer may not store or persist various encryption keys that are injected into the meters during the manufacturing process.

Abstract: Provided is an information processing apparatus including a binary random number generation unit configured to generate a binary random number string expressed with binary numbers of M bits (where M?2), and a ternary number string generation unit configured to generate a ternary number string by grouping the binary random number string in units of k bits and generating binary number strings of the k bits and by expressing the binary number strings of the k bits with ternary numbers of L symbols (where L is a maximum integer satisfying 3L?2M). The ternary number string generation unit generates the ternary number string by expressing a binary number string X of the k bits satisfying X?3L with the ternary numbers of the L symbols.

Abstract: A source device for transmitting content to a sink device is provided. The source device may include an interface configured to perform high-bandwidth digital content protection (HDCP) authentication with the sink device, and a controller configured to determine an HDCP version supported by the sink device, convert the content so as to be encrypted in the HDCP version supported by the sink device in response to a determination that another HDCP version applied to the content is not supported by the sink device, encrypt the converted content in the HDCP version supported by the sink device, and control the interface to transmit the content to the sink device.

Abstract: A data processing system includes a cryptographic processing module providing for cryptographic key generation. A method entails computing derived keys one time, during a first execution of a key generation process, such that they may subsequently be utilized for processing large quantities of data without being re-computed. The derived keys provide for the efficient cryptographic processing of data, including data frames.

Abstract: Technologies are generally described for proxy key generation, and signature generation and validation. In some examples, a cryptographic key generation system may include an original key generation unit configured to generate a public key and a private key for an original signer, a proxy key generation unit configured to generate one or more proxy public keys and one or more proxy private keys for one or more proxy signers, and a transmitter configured to transmit the proxy private keys respectively to the one or more proxy signers.

Abstract: A system and method of protecting, decrypting, and storing encryption keys. An encryption escrow module stores a library of indexed encryption algorithms. A keychain storage module includes a plurality of encrypted keys and/or keychains that are encrypted according to varying encryption algorithms of the encryption escrow module. Biometrics are used to index encrypted keychains to specific algorithms, but the two are kept separate. Since a naked key is never stored and only produced in cooperation with a specific user, the keychain storage module and the encryption escrow module, cracking attempts that compromise only two of the three groups are unable to generate any naked keys.

Abstract: An audio rendering device which uses multichannel speakers includes: a first delay computation unit which computes, based on rendering information about the multichannel speakers, a first delay corresponding to a primary wavefront which propagates in a predetermined traveling direction from a sound source that is each of speakers included in the multichannel speakers; a second delay computation unit which computes, based on the rendering information about the multichannel speakers, a second delay corresponding to a secondary wavefront which is generated by the primary wavefront and has a scattering wavefront; an addition unit which adds up the first delay and the second delay to compute a total delay; and a delay filter which applies the total delay to an input audio signal to generate a multichannel audio signal for use in rendering with the multichannel speakers, and outputs the multichannel audio signal to the multichannel speakers.

Abstract: Scaling, by a desired amount sm, the overall perceived loudness Lm of a multichannel audio signal, wherein perceived loudness is a nonlinear function of signal power P, by scaling the perceived loudness of each individual channel Lc by an amount substantially equal to the desired amount of scaling of the overall perceived loudness of all channels sm, subject to accuracy in calculations and the desired accuracy of the overall perceived loudness scaling sm. The perceived loudness of each individual channel may be scaled by changing the gain of each individual channel, wherein gain is a scaling of a channel's power. Optionally, in addition, the loudness scaling applied to each channel may be modified so as to reduce the difference between the actual overall loudness scaling and the desired amount of overall loudness scaling.

Abstract: An embodiment of the present invention provides a method for generating a downmixed signal, including: performing a time-frequency transform on a received left sound channel signal and a received right sound channel signal to obtain a frequency domain signal, and dividing the frequency domain signal into several frequency bands; calculating a sound channel energy ratio and a sound channel phase difference of each frequency band; calculating a phase difference between the downmixed signal and a first sound channel signal in each frequency band according to the sound channel energy ratio and the sound channel phase difference; and calculating a frequency domain downmixed signal according to the left sound channel signal, the right sound channel signal, and the phase difference between the downmixed signal and the first sound channel signal in each frequency band. This method effectively improves quality of stereo encoding and decoding.

Abstract: The present technique relates to a decoding device, a decoding method, an encoding device, an encoding method, and a program which can obtain a high-quality realistic sound. The encoding device stores speaker arrangement information in a comment region in a PCE of an encoded bit stream and stores a synchronous word and identification information in the comment region such that other public comments and the speaker arrangement information stored in the comment region can be distinguished from each other. When an encoded bit stream is decoded, it is determined whether the speaker arrangement information is stored on the basis of the synchronous word and the identification information stored in the comment region. Audio data included in the encoded bit stream is output according to the arrangement of the speakers corresponding to the determination result. The present technique can be applied to an encoding device.

Abstract: In a system and method for reproducing a sound field the orientation of a computing device may be detected. Several orientation indications may be used to detect the computing device orientation. The detected orientation may be relative to a sound field that is a spatial representation of an audible environment associated with the computing device. Audio transducers associated with the computing device may be selected in order to reproduce the sound field based on the detected orientation. A received encoding may be processed and reproduced with the selected audio transducers.

Abstract: In a system and method for encoding a sound field the orientation of a computing device may be detected. Several orientation indications may be used to detect the computing device orientation. The detected orientation may be relative to a sound field that is a spatial representation of an audible environment associated with the computing device. Microphones associated with the computing device may be selected in order to receive the sound field based on the detected orientation. The received sound field may be processed and encoded with associated descriptive information.

Abstract: In a system and method for maintaining the spatial stability of a sound field a balance gain may be calculated for two or more microphone signals. The balance gain may be associated with a spatial image in the sound field. Signal values may be calculated for each of the microphone. The signal values may be signal estimates or signal gains calculated to improve a characteristic of the microphone signals. The differences between the signal values associated with each microphone signal may be limited although some difference between signal values may be allowable. One or more microphone signals are adjusted responsive to the two or more balance gains and the signal gains to maintain the spatial stability of the sound field. The adjustments of one or more microphone signals may include mixing of two or more microphone. The signal gains are applied to the two or more microphone signals.

Abstract: Methods and apparatus to collect media identifying data are disclosed. An example method includes collecting audio data through a sound hole. The sound hole has a chamfered opening. The sound hole has an effective length-to-width ratio which is equal to or less than about 1.25 and analyzing the audio data to obtain media identifying data.

Abstract: A personal audio device, such as a wireless telephone, includes noise canceling circuit that adaptively generates an anti-noise signal from a reference microphone signal and injects the anti-noise signal into the speaker or other transducer output to cause cancellation of ambient audio sounds. An error microphone may also be provided proximate the speaker to measure the output of the transducer in order to control the adaptation of the anti-noise signal and to estimate an electro-acoustical path from the noise canceling circuit through the transducer. A processing circuit that performs the adaptive noise canceling (ANC) function also either adjusts the frequency response of the anti-noise signal with respect to the reference microphone signal, and/or by adjusting the response of the adaptive filter independent of the adaptation provided by the reference microphone signal.

Abstract: In a system and method for maintaining the spatial stability of a sound field a background noise estimate may be estimated for each of a first signal and a second signal. A first gain coefficient may be calculated responsive to the first audio signal and the background noise estimate of the first audio signal. A second gain coefficient may be calculated responsive to the second signal and the background noise estimate of the second signal. The gain coefficients may be calculated using one or more gain coefficient calculators. A common gain coefficient may be selected from one of the first gain coefficient and the second gain coefficient. The selected common gain coefficient may be one that results in a least amount of audio signal modification and may be applied to each of the first signal and the second signal.

Abstract: A system and method for mitigating microphone hiss may obtain a frequency spectrum characteristic for a microphone. A microphone that has limited dynamic range may create microphone hiss in an output signal. The microphone hiss may prevent a reproduction of a sound field, represented in an output signal of the microphone, from being perceived as a natural environment. The microphone frequency spectrum may be obtained using static measurements or calculated dynamically. A virtual noise floor may be calculated responsive to the microphone frequency spectrum and a desired noise floor. Gain coefficients may be calculated responsive to the output signal of the microphone. The gain coefficients may be calculated to mitigate undesirable signal content including background noise and echoes. The calculated gain coefficients may be modified responsive to the virtual noise floor. The modified gain coefficients may allow a reproduction of the sound field to be perceived as a natural environment.

Abstract: An earpiece set capable of emitting light and flashing according to changing of acoustic frequency including a pair of sound broadcasting units having earplugs and ear muffs, a pair of audio connectors and a pair of audio cables to transmit acoustic signals, wherein the audio cables electrically connect with a pair of control units for receiving acoustic signals, the control units are respectively electrically connected with a pair of LED light strips respectively extending along the audio cables to the sound broadcasting units, for driving the LED light strips for light emitting and flashing according to changing of acoustic frequency of the acoustic signals, thus effects of being easily discerned and providing safe warning and amusement can be achieved for a user.

Abstract: A microphone system has a package with a top, a bottom, and four sides that at least in part form an interior chamber. One of the sides forms an inlet aperture for communicating the inlet chamber with the exterior environment. The system also has first and second microphone dies, in a stacked relationship, respectively having a first and second diaphragms. A circuit die, positioned in electrical communication with the first and second microphone dies, is configured to mitigate vibrational noise from the first microphone die using a signal produced by the second microphone die or vice versa. The first and second microphone dies are positioned so that the first and second diaphragms are substantially the same distance from the inlet aperture in the side.

Abstract: An electronic device receives power through an earphone interface, and includes the earphone interface, a processing unit, an audio codec chip, a first switch and a second switch. The earphone interface includes a left channel pin, a right channel pin, a ground pin and a microphone pin. The first switch includes a first normally closed terminal, a first normally open terminal, a first common terminal and a first controlled terminal The second switch includes a second normally closed terminal, a second normally open terminal, a second common terminal and a second controlled terminal The processing unit includes a detect pin and a switch control pin. An earphone power line to power the electronic device is also provided.

Abstract: The present invention relates to a personal headphone device, system and method configured to combine a headphone and earbud into one acoustical product. The headphone earbud device is configured with an adaptive attachment for controlling the audio signals and/or device providing the audio signals. The adaptive connection includes an electro mechanical and/or mechanical connection so that when an earbud is placed into a headphone contact plates on the earbud mate with contact plates on the interior recess of the headphone so as to energize a loudspeaker in the headphone and or to control the device providing the audio signals.

Abstract: A mixed audio feed is generated in real-time from audio feeds of an event occurring within an event venue. The mixed audio feed is digitized and wirelessly streamed to an authorized mobile device located within the event venue.

Abstract: This disclosure enables a respective user to show support to a person of interest. One configuration includes a network (e.g., phone network, Internet, etc.) configured to receive input from a respective user (e.g., family member, friend, etc.) and control a remotely located device. The remote device can be located at any suitable locations such as a gravesite of a deceased party, a hospital room in which a disabled patient resides, etc. Via input from a respective user, the respective user can control different functions of a remote target device such as audibly communicate one or more messages in a vicinity of the gravesite, control a remote device such as light source (such as a candle, light emitting diode, etc.), etc.

Abstract: A method of operating an audio entertainment system in a vehicle includes detecting an operating parameter of the vehicle. A sensor signal corresponding to the detected operating parameter is transmitted. Music that is audibly produced within the vehicle is musically modified dependent upon the sensor signal.

Abstract: The present invention provides a loudspeaker with at least two diaphragm assemblies, in which the reactive forces acting on the magnets in each assembly cancel. By linking the two magnets together via a non-rigid material, resonance in the drive assemblies is highly damped and corresponding distortion of the loudspeaker output is reduced.

Abstract: Provided is an audio output device capable of preventing noises and improving the S/N ratio even if no audio signal is inputted in the middle of the input of audio signals, or even if an audio-signal input state and a no-signal state are alternately repeated. In the provided audio output device, a multiplier is provided on the input side of each of the delayers. Each multiplier multiplies the addition output of the corresponding one of adders by a multiplier coefficient supplied by the coefficient counter. If there is no input of digital audio signals into a ?? modulator, the counter control circuit decreases the output of the coefficient counter down to 0 stepwise at predetermined intervals.

Abstract: A system and method for audio bandwidth dependent noise suppression may detect the audio bandwidth of an audio signal responsive to one or more audio indicators. The audio indicators may include the audio sampling rate and characteristics of an associated compression format. Noise suppression gains may be calculated responsive to the audio signal. Noise suppression gains may mitigate undesirable noise in the reproduced output signal. The noise suppression gains may be modified responsive to the detected audio bandwidth. Less noise reduction may be desirable when more audio bandwidth is available. The modified noise suppression gains may be applied to the audio signal.

Abstract: A noise estimation control system may limit increases of a stored background noise estimate in response to a detected noise feedback situation. The system receives an input audio signal detected within a space, and a reference audio signal that is transmitted by a speaker as an aural signal into the space. A signal processor processes the input audio signal and the reference audio signal to determine a coherence value based on an amount of the aural signal that is included in the input audio signal. The signal processor also calculates an amount to adjust the stored background noise estimate based on the coherence value and a determined background noise level of the input audio signal.

Abstract: A method for dynamically adjusting the spectral content of an audio signal, which increases the harmonic content of said audio signal, said method comprising translating an encoded digital signal into data bands, creating a psychoacoustic model to identify sections of said data bands that are deficient in harmonic quality, analyzing the fundamental frequency and amplitude of said harmonically deficient data bands, creating additional higher order harmonics for said harmonically deficient data bands, adding said higher order harmonics back to said encoded digital signal to form a newly enhanced signal, inverse filtering said newly enhanced signal, and converting said inverse filtered signal to an analog waveform for consumption by the listener.

Abstract: The present invention relates to a method for dynamically adjusting a gain of parametric equalizer according to an input signal, a dynamic parametric equalizer employing the same and a dynamic parametric equalizer system employing the same wherein a gain of parametric equalizer is dynamically adjusted according to a level of an input digital audio signal to prevent distortion of output signal.

Abstract: A mobile terminal and controlling method thereof are disclosed, which facilitates a terminal to be used in further consideration of user's convenience. The present invention includes a display unit configured to display an image signal of a video, an audio output unit configured to output an audio signal of the video, and a controller configured to separate the audio signal of the video into sound sources, and control a volume of at least one of the sound sources manually or automatically.

Abstract: The invention provides earphone arrangements for use with electronic host devices, such as cellular telephone handsets and music storage and reproduction devices, and it provides in particular earphone arrangements with ambient noise cancellation (“ANC”). Typically, in prior arrangements, the complex signal processing required for ANC is carried out in the earphones themselves and/or in a pod connected between the earphones and the host device. The invention configures the processing elements such that the ANC signal-processing function is transferred to an ANC processor incorporated into the host device.

Abstract: A hearing loss compensation apparatus and method compensate for distortion caused by a change in a sound transmission path using hearing characteristics of a user. The hearing loss compensation apparatus may include a sound direction detection device configured to detect a sound generation direction, in which a sound generates, using one or more microphones, and a sound compensation device configured to compensate the sound using hearing characteristics of a user corresponding to the sound generation direction. Additionally, a hearing characteristics measurement apparatus and method provide a way to obtain such hearing characteristics information.

Abstract: A method of fitting a hearing aid connected to a mobile terminal is provided. The method may include acquiring location information of a location of the mobile terminal, transmitting the location information to a server, receiving, from the server, information about at least one recommended fitting parameter model adapted to hearing conditions of the location and hearing characteristics of a user of the hearing aid, and displaying the information about the at least one recommended fitting parameter model. The method may also include receiving a fitting parameter model selected by the user from among the at least one recommended fitting parameter model, and controlling the hearing aid to function according to the received fitting parameter model.

Abstract: Disclosed in some examples is a method of authentication, the method comprising monitoring for the presence of a hearing instrument; responsive to detecting the hearing instrument: establishing a short-range wireless communication session with the hearing instrument; determining an identity of the hearing instrument wearer; and responsive to determining the identity of the wearer, setting a value indicating that the hearing instrument wearer is authenticated with and within the wireless range of the presence device.

Abstract: A sealed enclosure providing a microphone contained therein extended protection from the weather and some other outdoor hazards while not substantially altering the dynamic range of said microphone. The enclosure consists of a layer of liquid-water impermeable fabric, a layer of acoustical foam, a protective cage, and a wind noise dampening external cover. The enclosure is assembled around a microphone and a commercial foam windscreen, or around a custom high-wind foam windscreen first assembled around the microphone. As such, the weatherproof enclosure enables continuous use of boundary microphones and many other types and styles of microphones outdoors year round while supplementing the wind noise reduction of the enclosed foam windscreen.