Abstract: A wireless audio system using low overhead in-band control and audio transmission is provided. The wireless audio system includes a first wireless audio device configured to operate separate physical layer channels for audio data and control data, and transmit the audio data and control data using a single wideband carrier. The wireless audio system also includes one or more second wireless audio devices configured to receive the audio data and control data, and execute an instruction based on the control data.

Abstract: Methods, systems, and devices for wireless communications are described. Generally, a user equipment (UE) may transmit an indication to a base station that a total received power for the UE exceeds an overload threshold. In some cases, the indication may be an explicit request for a decrease in transmit power by a specific amount of reduction. The UE may transmit indications that the overload threshold is exceeded to multiple transmit/receive points. The indications may request specific transmit power reduction amounts that are specific to the TRPs, or may be generic indications that the overload threshold is exceeded, and the TRPs may coordinate to reduce their respective transmit powers to satisfy the overload threshold at the UE. The base station may allocate a first set of resources for near field UEs and a second set of resources for far field UEs.

Abstract: The present disclosure pertains to a line-powered current measurement system to mount to an electrical conductor and related methods. In one embodiment, a system may comprise a current transformer to electrically couple to an electrical conductor and to generate a secondary current proportional to a primary current in the conductor. A power harvesting subsystem may harvest power from the secondary current in a first configuration. A switching subsystem may transition the line-powered current measurement device between the first configuration and a second configuration, in which a current measurement subsystem generates a measurement of the secondary current. The switching subsystem may provide the secondary current to the power harvesting subsystem in the first configuration and may bypasses the power harvesting subsystem in the second configuration. A communication subsystem may transmit the measurement of the secondary current to a receiver device.

Abstract: A method, apparatus and computer program product are disclosed. The method includes acquiring an instruction, picking up an audio signal with an electronic device, determining whether an output apparatus is connected to the electronic device, outputting the audio signal via the output apparatus, and recording the audio signal. The apparatus includes a processor, an output apparatus, a microphone, and a storage apparatus, wherein the processor acquires an instruction, picks up an audio signal, determines whether the output apparatus is connected, outputs the audio signal via the output apparatus and records the audio signal to the storage medium. The computer program product includes executable code to perform acquisition of an instruction, pickup of an audio signal with an electronic device, a determination of whether an output apparatus is connected, output of the audio signal, and recording of the audio signal.

Abstract: A system and method for coupling a communications device to a primary communications network having a first communications format and to a secondary communications network having a second communications format. The system and method can comprise a first port configured for connecting to the communications device, a second port configured for connecting to the primary communications network and facilitating the communication of the data between the primary communications network and the communications device through the first port.

Abstract: Embodiments relate generally to devices, systems, and methods for assessing a hearing protection device. A method for assessing a hearing protection device may comprise monitoring sounds with a hearing protection device; transmitting sound measurements from the hearing protection device to a portable electronic device; transmitting the sound measurements from the portable electronic device to a server; comparing, with the portable electronic device or the server, sound measurements to a look-up table; suggesting, with the portable electronic device, the most suitable hearing protection device to be worn based on a comparison between the sound measurements and the look-up table.

Abstract: An audio reproduction apparatus includes: a signal processing unit that converts an audio signal into N channel signals, where N is an integer greater than or equal to 3; and a speaker array including N speaker elements that respectively output the N channel signals as reproduced sound, wherein the signal processing unit includes: a beam formation unit that performs a beam formation process of resonating the reproduced sound output from the speaker array at a position of one ear of the listener; and a cancellation unit that performs a cancellation process of preventing the reproduced sound output from the speaker array from reaching a position of the other ear of the listener.

Abstract: A system and method are provided for adaptively processing audio commands supplied by a user in an aircraft cabin, and includes receiving ambient noise in the aircraft cabin via one or more audio input device, sampling, with a processor, the received ambient noise, and analyzing, in the processor, the sampled ambient noise and, based on the analysis, selecting one or more filter functions and adjusting one or more filter parameters associated with the one or more selected filter functions. Audio and ambient noise are selectively received via the one or more audio input devices, and are filtered, through the selected one or more filter functions, to thereby supply filtered audio.

Abstract: An in-ear noise dosimeter in the form of an earplug which senses sound in the ear canal using an eartip which has a sound delivery channel that couples sound at the end closest to the eardrum to an earplug microphone. The earplug can communicate wirelessly with a remote data collection and processing system. A dock unit for storing the earplugs when not worn can compensate for differences in unoccluded-ear versus occluded-ear responses by an acoustic compensator. An electronic compensation filter can be modified by a proximity switch in the earplug which changes state when the earplug is worn in the ear versus stored in a dock unit. The dosimeter can also have a temperature sensor for sensing human body temperature and remotely-located wireless LEDs used to alert the user of high noise dosage. Data can also be downloaded from the earplug using a reader unit.

Abstract: Various embodiments of systems, computer program products, and methods to provide instant translation of user interfaces of a web application are described herein. In an aspect, a request is received to translate a user interfaces of a web application from a source language to a target language. A translatable user interface element associated with the user interface of the web application is determined. Further, the translatable user interface element is translated from the source language to the target language based on dynamic metadata associated with the translatable user interface element. The translated translatable user interface element in the user interface of the web application is rendered.

Abstract: Embodiments include a system, method, and computer program product for providing seamless PTT services to a user. In an embodiment, a push-to-talk (PTT) component provides, to the user, PTT services with a talk group through a connectivity to a data channel in a mobile network. A voice monitoring component monitors attributes of data connectivity on the data channel to determine when a voice quality falls below a threshold. When the voice quality is determined to fall below the threshold, a connectivity component requests the user to switch the connectivity to a voice channel. Then, the connectivity component dials a telephone number through the voice channel to access a voice conference bridge that provides the PTT services. To provide the PTT services, the voice conference bridge sets up a routing path to a session initiation protocol session for the talk group, and routes voice communication through the SIP session.

Abstract: There is provided a signal processing apparatus advantageous in terms of sound source separation performance. The signal processing apparatus includes a dividing unit configured to divide audio signal acquired by a plurality of audio acquisition units into components of a plurality of different frequency bands, and a processing unit configured to form, based on the audio signal, a plurality of directional beams having different directivities in accordance with a target direction and a target width. Each of the plurality of directional beams has directivities in different directions for the respective components of the frequency bands divided by the dividing unit.

Abstract: An augmented reality environment allows interaction between virtual and real objects. Beamforming techniques are applied to signals acquired by an array of microphones to allow for simultaneous spatial tracking and signal acquisition from multiple users. Localization information such as from other sensors in the environment may be used to select a particular set of beamformer coefficients and resulting beampattern focused on a signal source. Alternately, a series of beampatterns may be used iteratively to localize the signal source in a computationally efficient fashion. The beamformer coefficients may be pre-computed.

Abstract: Power management is provided in an Internet protocol (IP) telephone and system to provide energy savings during times that the IP telephone is not in use or use is not expected. A low-power operating mode disables at least a portion of the IP telephone. The low-power operating mode may be initiated by a command received by the IP telephone from the IP telephone controller according to a schedule, which may be modified locally by the user to individualize the user's schedule. The low-power operating mode may alternatively be activated manually by a user pressing a special key, sequence or combination. The low-power operating mode is canceled upon an indication that a user either is or should be present at the IP telephone.

Abstract: Methods and systems to produce audio output signals from audio input signals. In one embodiment, a first portion of the audio input signals can be pre-processed, with the output used to modulate ultrasonic carrier signals, thereby producing modulated ultrasonic signals. The modulated ultrasonic signals can be transformed into a first portion of the audio output signals, which is directional. Based on a second portion of the audio input signals, a standard audio speaker can output a second portion of the audio output signals. Another embodiment further produces distortion compensated signals based on the pre-processed signals. The distortion compensated signals can be subtracted from the second portion of the audio input signals to generate inputs for the standard audio speaker to output the second portion of the audio output signals. In yet another embodiment, noise can be added during pre-processing of the first portion of the audio input signals.

Abstract: A device is disclosed, the device comprising a least one microphone, at least one output audio transducer driver, at least one processor, a storage comprising a set of instructions executed locally, wherein the set of instructions causes the processor to: continuously measure an ambient sound pressure level when the device is an on state, calculate a sound pressure level of the output audio transducer driver when the output audio transducer is active, combine the two sound pressure levels into a total sound pressure level.

Abstract: A system and method for coupling a communications device to a primary communications network having a first communications format and to a secondary communications network having a second communications format. The system and method can comprise a first port configured for connecting to the communications device, a second port configured for connecting to the primary communications network and facilitating the communication of the data between the primary communications network and the communications device through the first port.

Abstract: Provided are a VOIP terminal and a ringing parameter setting method therefor. An application scenario detecting module of the VOIP terminal detects an application scenario of the VOIP terminal, and sends a detection result to a parameter matching module of the VOIP terminal, the parameter matching module matches a ringing parameter according to the detection result, and then a parameter setting module of the VOIP terminal sets the VOIP terminal according to the ringing parameter obtained by the matching. Manual setting according to experience is not involved. The set ringing parameter can better meet requirements of various voice user terminals under the VOIP terminal compared with an existing manual setting manner. Abnormal conditions that some voice user terminals ring briefly all the time, some voice user terminals are muted after ringing a sound, even some voice user terminals have no ringing all the time, and so on under the same VOIP terminal can be reduced, and user experience can be improved.

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 wireless ear phone and a sound broadcasting method thereof are provided. The wireless ear phone includes a first signal sensor, an electromagnetic signal transmitter, a second signal sensor and a controller. The first signal sensor generates an environment light detection result by detecting an environment light. The electromagnetic signal transmitter transmits an interruptible electromagnetic signal. The second signal sensor is disposed on a transmission path of the interruptible electromagnetic signal, and the second signal sensor generates an electromagnetic signal detection result by detecting whether the interruptible electromagnetic signal or the environment light is received. The controller enables or disables an operation of sound broadcasting of the wireless ear phone according to the electromagnetic signal detection result. When the wireless ear phone is put into an ear of a user, the transmission path is cut-off by the ear of the user.

Abstract: A voice agent device includes: a position detection unit which detects a position of a person in a conversation space to which the voice agent device is capable of providing information; a voice volume detection unit which detects a voice volume of the person from a sound signal in the conversation space obtained by a sound acquisition unit; a conversation area determination unit which determines a conversation area as a first area including the position when the voice volume has a first voice volume value and determines the conversation area as a second area including the position and being smaller than the first area when the voice volume has a second voice volume value smaller than the first voice volume value, the conversation area being a spatial range where an utterance of the person can be heard; and an information provision unit which provides provision information to the conversation area.

Abstract: Audio clipping is prevented by attenuating an audio signal in accordance with values retrieved from a gain table. Corresponding amplitude values of a stereo audio signal are evaluated to determine a maximum of the values. The amount by which the maximum exceeds a predetermined threshold is used to calculate a table index, which is used to retrieve a gain value from the gain table. The gain value is then applied to the audio signal. The gain table is configured so that increasing index values produce decreasing gain values.

Abstract: There is provided a signal processing apparatus including a recognition section configured to recognize at least one of a state of a first user, a state of a second user, and a relationship between the first user and the second user, a signal processing unit configured to perform signal processing on voice information corresponding to voice articulated by the first user, or voice information corresponding to voice that is transmitted from a remote location and articulated by the second user, and a control unit configured to control the signal processing unit to perform signal processing on at least one of the voice information corresponding to the voice articulated by the first user, and the voice information corresponding to the voice that is transmitted from the remote location and articulated by the second user, in accordance with a recognition result of the recognition section.

Abstract: A method and apparatus related to spatial filtering to select target terminals only located at a predetermined distance among a plurality of target terminals, using a propagation delay between an electromagnetic signal and an acoustic signal are provided. A user terminal transmits the acoustic signal to neighboring terminals to select terminals of which distances from the user terminal correspond to a distance from the user terminal to a target terminal among neighboring terminals. The electromagnetic signal is transmitted to the neighboring terminals in a predetermined transmission delay. Response signals are received from neighboring terminals of which distances from the user terminal are within a predetermined range, among the neighboring terminals that have received the acoustic signal and the electromagnetic signal. The neighboring terminals receive the acoustic signal from the user terminal, and receive the electromagnetic signal transmitted in the transmission delay from the user terminal.

Abstract: Features are disclosed for estimating a noise level using a variable step size. An acoustic echo canceller (AEC) may be configured to perform echo cancellation. The acoustic echo canceller may determine an estimated echo using a playback signal. The acoustic echo canceller also may determine an estimated error using the estimated echo and a microphone signal. A variable step size may be determined using the estimated error and the microphone signal. Noise reduction may be performed using the variable step size.

Abstract: The present invention is applicable to the field of communications technologies, and provides a method and apparatus for controlling a call audio channel on a communications terminal, and a communications terminal. In the method, first, a voice signal received by a microphone of audio devices connected to a communications terminal is detected, where the audio devices include an internal audio device and at least one external audio device; then, an audio device with a received voice signal meeting a preset condition is selected as a target audio device; finally, the target audio device is set as a current call audio channel. In this way, a user neither needs to remember an audio device that has been connected nor needs to manually switch to a desired call audio channel, which brings convenience to the user.

Abstract: Unshielded twisted pair (UTP) termination circuits, UTP termination circuit systems, and methods of terminating an UTP circuit are provided. UTP termination circuits can include a plurality of resistors in parallel, where each of the plurality of resistors is configured to be coupled to one of a plurality of unshielded twisted pairs of an UTP cable, a shunt resistor is coupled in series with the plurality of parallel resistors, and a capacitor is connected to ground in series with the shunt resistor.

Abstract: Methods and apparatus, including computer program products, implementing and using techniques for translating digital content from a source language to a target language. A message is displayed to a user. The message contains digital content to be translated from the source language to the target language, as well as the context of the digital content in the source language and/or a reference to a context in which the digital content occurs. A proposed translation of the digital content into the target language is received from the user. The proposed translation is submitted to a translation server.

Abstract: In one embodiment, a low cost, simple circuit for detecting an off-hook condition of a telecommunication line comprising tip and ring signal lines is provided. The circuit comprises a voltage divider for coupling between the tip and ring lines without an intervening transistor and having a node at which is presented a scaled version of a voltage across the voltage divider. The circuit further comprises a transistor having a control terminal coupled to the node, a first current flow terminal coupled to a voltage source, and a second current flow terminal coupled to an output terminal, wherein the output terminal bears a value that is indicative of a voltage across the tip and ring lines and thus whether the telecommunication line is off-hook.

Abstract: A self configuring headset system comprising an interface unit, a method of setting up a headset system to a telephone device using and interface unit and a telephone interface unit is provided.

Abstract: One feature discloses a method operational at a multimedia mobile call transfer device, the method comprising: receiving a first video data associated with a multimedia call from a mobile device; receiving a first audio data associated with the multimedia call from the mobile device; processing the received first video data and the received first audio data; transmitting the processed first video data to a multimedia device for display on the multimedia device; and transmitting the processed first audio data to the multimedia device for play on speakers of the multimedia device.

Abstract: An audio conversation apparatus includes an assignment section for individually assigning units of spatial information which are different from each other, either to parties-to-talk-with each belonging to one of a plurality of predetermined groups, respectively, or to the plurality of predetermined groups, respectively; and a localization section for localizing, in accordance with the units of spatial information assigned by the assignment section, audio data transmitted from outside, and one of a reproduction section connected to the audio conversation apparatus and a reproduction section included in the audio conversation apparatus outputs an audio in accordance with the audio data having been localized by the localization section.

Abstract: A communications system includes a communications device having a plurality of access modules each having a port and connected to a communications line and a plurality of transmitters with the respective transmitter associated in one-to-one correspondence with the communications line of an access module. Each transmitter has a line driver and is configured to couple communications signals to a respective communications line. A voltage source is connected to the line drivers and configured to provide a bias voltage to the line drivers that varies depending on a selected minimum power level. A controller is connected to the voltage source and has logic configured to change the bias voltage to the line drivers. The controller is responsive to a minimum data rate for each bias voltage.

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: The location of a mobile terminal associated with a user is determined using GPS or by monitoring signals generated and received by the mobile terminal, etc. Coordinates for at least one hot spot defined by the user are accessed to determine whether the location of the mobile terminal is within the hot spot. When an incoming contact is received for the user, a determination is made as to whether there are any contact routing rules associates with the user. If so, the record of contact routing rules associated with the user is accessed. If the record of contact routing rules includes at least one contact routing rule assigned to a hot spot, the contact is routed according to the at least one contact routing rule, based on whether or not the mobile terminal is determined to be within the hot spot.

Abstract: A method for automatically configuring an interface unit (112, 212, 912, 1012, 1112) for connecting a headset (110, 210, 910, 1110) and a telephone (100) by electrical connections (992, 993, 1092, 1093). The electrical connections (992, 993, 1092, 1093) between the interface unit (112, 212, 912, 1012, 1112) and the telephone (100) comprise two pairs of electrical paths having a number of distinct permutation settings. Each of the number of distinct permutation settings have a complementary setting together forming a number of distinct complementary setting pairs. The method comprises the steps of: a) receiving a test signal, b) for each distinct permutation setting generating an output signal in response to the test signal, c) determining a least preferred distinct permutation setting based on the output signals, d) selecting the complementary setting to the least preferred distinct permutation setting as a preferred setting.

Abstract: Embodiments of the present invention disclose a telephone terminal. The telephone terminal includes: a first handle end, a second handle end, a voice processing unit, and an input-output unit. The voice processing unit receives a first voice signal sent by the first handle end and a second voice signal sent by the second handle end, processes the first voice signal and the second voice signal, and sends the processed voice signal to a communication peer through the input-output unit. According to the embodiments of the present invention, voice signals of the two handle ends are received, processed, and then sent to the communication peer, a problem of failing to send a speech signal of a user to the communication peer when the user holds the handle upside down may be solved, and user experience is improved.

Abstract: A circuit apparatus and a method for detecting an earphone in a portable terminal are provided. The apparatus includes an earjack, a first switch, a second switch, a comparator, and a main chip. The earjack outputs a signal informing whether an earphone is inserted, provides a signal input from a third pole of the inserted earphone to the first switch, and provides a signal input from a fourth pole to the second switch. The first switch outputs a signal provided from the earjack to one of a comparator and a ground depending on an input control signal. The second switch outputs a signal provided from the earjack to one of the comparator and the ground depending on the input control signal. The comparator compares a voltage of a signal provided from one of the first switch and the second switch with a reference voltage to output a signal representing a comparison result.

Abstract: Methods and apparatus, including computer program products, implementing and using techniques for translating digital content from a source language to a target language. A message is displayed to a user. The message contains digital content to be translated from the source language to the target language, as well as the context of the digital content in the source language and/or a reference to a context in which the digital content occurs. A proposed translation of the digital content into the target language is received from the user. The proposed translation is submitted to a translation server.

Abstract: A method and system automatically activates a mobile station in a wireless communications network. The system includes an over the air activation function (OTAF) processor in the network that initiates an activation process in response to receiving a registration message from a mobile switching center serving the mobile station requesting the activation. Each mobile station has a unit of information stored into it at the time of its manufacture to enable it to request over the air activation. That unit of information is either the network routing address of the OTAF processor, or alternately, it is a value that is translatable into that address, either an OTAF ID number that is the same value for every mobile station or it is a sequentially serialized dummy value for the mobile identification number (a dummy MIN).

Abstract: An automatic sidetone controller for a digital telecom device, such as a headset or head-set adapter having a transmit (TX) channel, a receive (RX) channel, and circuitry for generating a sidetone signal by coupling a portion of a signal on the TX channel onto the RX channel via a sidetone path, is operable to detect when the TX channel of the device is active and inactive and to respectively activate and deactivate the sidetone path in response thereto without need for manual intervention by the user.

Abstract: A mobile communication device and a method of setting tone color, which allow a user to set the tone color of received sound. Provided are a normal mode, which sets the equalizer using GCF standards stored in an internal memory or equalizer setting values selected by a provider, a country-specific mode, which uses country-specific setting, and a user mode, in which a user can set frequency-specific gains of the received sound, and one mode is selected from the provided mode, so that the tone color of the received sound can be adjusted according to the selection. Telephone speech quality can be optimized for user preference, network environments and language characteristics.

Abstract: A video conferencing system contains one or more display adjusting components, whereby an object to be displayed can be adjusted to appropriately fit various sized display screens. A display adjusting component is contained within the sending client, which adjusts the image of the object to be appropriately displayed to one or more receiving clients. The receiving clients also contain a display adjusting component, which can further adjust the image of the object to be displayed, as necessary. The multimedia conferencing server of the video conferencing system also contains a display adjusting component, which negotiates parameters of the sending and receiving clients. Any of the display adjusting components can function alone, or in any combination together. A method, and computer readable media which contain computer readable instructions to perform a method, of adjusting an image for video conference display are also described.

Abstract: Seamless wideband support is afforded by utilizing split-band data streams. For wideband signals, the 8 kHz bandwidth is divided into a low band, with approximately 0-4 kHz bandwidth, and a high band, with approximately 4-8 kHz bandwidth. Narrowband functions and services operate on the low band, while wideband functions and services operate on both low and high bands.

Abstract: A power supply circuit including a voltage regulator and a headphone driving circuit employing the same are disclosed. In accordance with the present invention, the power supply circuit may provide stable power supply to the amplifier since the power supply circuit generates the constant voltage +Vreg and the constant voltage ?Vreg from +VDD robust to stronger than a power noise, and the balanced signal path for isolating the noise of the input signal to be amplified is not required, thereby simplifying the constitution of the circuit.

Abstract: Methods and apparatus, including computer program products, implementing and using techniques for translating digital content from a source language to a target language. A message is displayed to a user. The message contains digital content to be translated from the source language to the target language, as well as the context of the digital content in the source language and/or a reference to a context in which the digital content occurs. A proposed translation of the digital content into the target language is received from the user. The proposed translation is submitted to a translation server.

Abstract: A unified communications appliance provides integration of various types of information, regardless of the modality, in a common, centralized interface where the various types of information are grouped based on what they are related to. For example, as is common with most modalities of information exchange, there exists a “subject” that is present in one of the fields of communication. The ability to associate all of the types of communication with a common “subject” (or conversation) and provide an interface that allows access to the various types of information, regardless of the modality is provided by the unified communications interface.

Abstract: In one embodiment, a low cost, simple circuit for detecting an off-hook condition of a telecommunication line comprising tip and ring signal lines is provided. The circuit comprises a voltage divider for coupling between the tip and ring lines without an intervening transistor and having a node at which is presented a scaled version of a voltage across the voltage divider. The circuit further comprises a transistor having a control terminal coupled to the node, a first current flow terminal coupled to a voltage source, and a second current flow terminal coupled to an output terminal, wherein the output terminal bears a value that is indicative of a voltage across the tip and ring lines and thus whether the telecommunication line is off-hook.

Abstract: An audio host device has an electrical interface having a speaker contact, a microphone contact, and a reference contact. The reference contact is shared by a microphone and a speaker. The reference contact is also directly coupled to a power return plane of the audio host device. A difference amplifier is provided, having a cold input and a hot input. The hot input is coupled to the microphone contact. A switched attenuator circuit is also provided that has first and second states. In the first state, the attenuator circuit couples a sense point of the reference contact to the cold input, while in the second state the cold input is isolated from the reference sense point. Other embodiments are also described and claimed.

Abstract: An audio host device has a connector having a speaker contact, a microphone contact, and a reference contact. The reference contact is shared by a microphone and a speaker. The reference contact is also directly coupled to a power return plane of the audio host device. A difference amplifier is provided, having a first cold input, a second cold input, and hot input. The first cold input is coupled to an output of the speaker amplifier, while the second cold input is coupled a sense point for the reference contact. The hot input is coupled to the microphone contact. Other embodiments are also described and claimed.