Abstract: A method and system to manage connectivity of a client device in a mesh network including one or more network devices. A first network device determines a first client link cost metric associated with a first connection between the first network device and the client device. The first network device determines a first mesh link cost metric associated with a second connection between the first network device and a destination device. A first total connection cost metric is determined based on the first client link cost metric and the first mesh link cost metric. The first network device identifies a set of total connection cost metrics associated with the one or more network devices and destination device of the network. The first network device executes a connection decision in response to a comparison of the set of total connection cost metrics.

Abstract: A communication device for allocating tones is described that includes a processor and instructions in memory in electronic communication with the processor. The communication device determines whether a bandwidth for signal transmission is 20, 40, 80 or 160 megahertz (MHz). The communication device respectively allocates tones for 20, 40, 80 or 160 MHz as follows: for a very high throughput (VHT) signal A1 (VHT-SIG-A1): 52, 104, 208, 416; a VHT signal A2 (VHT-SIG-A2): 52, 104, 208, 416; a VHT short training field (VHT-STF): 12, 24, 48, 48; one or more VHT long training field(s) (VHT-LTF(s)): 56, 114, 242, 484; a VHT signal B (VHT-SIG-B): 56, 114, 242, 484; and a data field (DATA): 56, 114, 242, 484. The communication device also transmits the signal.

Abstract: A method of operation beamforms a plurality of microphone outputs to obtain a plurality of virtual microphone audio channels with at least one audio output channel and at least one audio control channel. The method performs voice recognition on the audio control channel to detect voice commands for controlling audio output channel attributes, and adjusts an audio channel attribute in response to detecting a voice command. Adjusting an attribute of the audio channel may be accomplished by, for example, controlling one or more parameters of an adjustable beamformer. The detected voice commands for controlling audio channel attributes may include voice commands for controlling audio sensitivity zooming, panning in a specified direction, focusing on a specified direction, blocking a specified direction, mixing a narrator's voice, blocking a narrator's voice, or reducing background noise. An apparatus that performs the method of operation is also disclosed.

Abstract: The present disclosure discloses a method, system, and device for forwarding data frames, and relates to the communication field. The method includes: An upstream near-end device in a faulty part of a working tunnel activates a segmental protection tunnel after receiving a data frame when a part of the working tunnel fails, where the start node of the segmental protection tunnel is the upstream near-end device and the end node of the segmental protection tunnel is a downstream near-end device in the faulty part; the upstream near-end device forwards the data frame to the downstream near-end device through the segmental protection tunnel; and the data frame at the downstream near-end device is forwarded to the destination device through the faultless part of the working tunnel. The present disclosure provides segmental protection for the PBB TE tunnel.

Abstract: In an embodiment of the present invention, a programmable matrix is provided for flexibly allocating communication resources among functional circuit blocks. For example, in an embodiment of the present invention, a programmable matrix is provided that allocations communications channels such as transceiver channels among various PCIe hard IP blocks that may be contained within a programmable logic device (PLD).

Abstract: Using switching technologies to duplicate packets of a digital stream (e.g., digital video stream) sent from one workstation to multiple recipient workstations, where the switching technologies enable the multiple streams sent from the switch to the recipient workstations to be generated from a single digital stream sent from the sending workstation to the switch. Data units, such as video data units, may be transmitted by using a switch to enable receipt of a stream of data units including a payload portion and an attribute portion from at least two conferencing participants. The switch is used to duplicate at least a subportion of the payload portion of a data unit within the stream of data units, and to enable access to the duplicated subportion of the data unit by two or more conferencing participants.

Abstract: Systems and methods are disclosed for selecting resources for direct device to device (D2D) communications in a cellular communication network. In one embodiment, a downlink resource is selected as a resource for a direct D2D communication link between a first wireless device and a second wireless device if a base station serving each of the first and second wireless devices is equipped with an interference cancellation receiver and both the first and second wireless devices are less than a predefined threshold radio distance from their serving base station. An uplink resource is selected as a resource for the direct D2D communication link if the base station serving each of the first and second wireless devices is equipped with an interference cancellation receiver and at least one of the first and second wireless devices is more than the predefined threshold radio distance from its serving base station.

Abstract: An apparatus comprising an ingress port configured to receive a signal comprising a plurality of encoded audio signals corresponding to a plurality of sources; and a processor coupled to the ingress port and configured to calculate a parameter for each of the plurality of encoded audio signals, wherein each parameter is calculated without decoding any of the encoded audio signals, select some, but not all, of the plurality of encoded audio signals according to the parameter for each of the encoded audio signals, decode the selected signals to generate a plurality of decoded audio signals, and combine the plurality of decoded audio signals into a first audio signal.

Abstract: An apparatus for removing an echo(es) from audio content may include a processor and memory storing executable computer code causing the apparatus to at least perform operations including receiving combined audio content including voice data associated with speech of users in a call and information including audio data provided to the users. The computer program code may further cause the apparatus to remove a first echo of a first item of voice data associated with a user(s), from the combined audio content, based in part on a prior detection of the first item of voice data. The computer program code may further cause the apparatus to remove a second echo of the audio data, from the combined audio content, based in part on a previous detection of the audio data or a previous detection of data corresponding to the audio data. Corresponding methods and computer program products are also provided.

Abstract: The illustrative embodiments described herein provide improved systems and methods for conferencing enterprise and non-enterprise callers. In one embodiment, a method includes receiving a first call initiated by an enterprise caller at an enterprise, communicating with an enterprise media server to initiate a tributary conference on the enterprise media server, joining the enterprise caller to the tributary conference such that the enterprise caller is in data communication with the tributary conference, receiving a second call initiated by a non-enterprise caller via a public switched telephone network, interfacing the non-enterprise caller with a data center media server, creating a home conference on the data center media center, joining the second call to the home conference, and interfacing the home conference on the data center media server to the tributary conference on the enterprise media server to form a linked conference between the enterprise caller and the non-enterprise caller.

Abstract: Techniques for determining whether a cellular device is suspect, i.e., perhaps serving as an activator for a device such as a bomb. One way of doing this with cellular telephones that are in the idle state is to use a baiting beacon to bait and automatically call all the cellular telephones in an area that are in the idle state. If the call to a given cellular telephone is not answered by a human voice, the cellular telephone is suspect. Another way of doing this with cellular telephones that are in the traffic state is to use surgical analysis to examine the DTX pattern for the telephone. If it indicates persistent silence, the cellular telephone is suspect. The surgical analysis may also be used to trace the DTX pattern back to another telephone that is controlling the suspect cellular device.

Abstract: An apparatus includes a transmitter configured to transmit data via a plurality of communication channels of a communication connection. The transmitter is further configured to transmit a same sequence for vector training on each of the communication channels. The sequences for at least two of the communication channels are shifted with respect to each other.

Abstract: In one aspect, audio streams are added to a mix until the mix is either complete (i.e., all audio streams have been added) or the mix is closed early (i.e., before the mix is complete). In another aspect, audio and video streams are synchronized by playing back the audio stream and then synchronizing display of the video frames to the playback of the audio stream.

Abstract: A method of automatically establishing an input language for a handheld electronic device includes receiving a request to initiate the composition of a new message, receiving a selection of a particular contact that is the intended recipient of the message, and determining whether the particular contact has a stored preferred input language. If a preferred input language is stored, the method includes determining whether the preferred input language is different than the current input language (which may be a pre-established default input language). If the preferred input language is different, the method further includes switching the current input language to the preferred input language. Also provided is an improved handheld electronic device having one or more routines adapted to implement the method. Also provided is a method of indicating a preferred language in association with a call received by or placed to the device.

Abstract: A new approach is proposed that contemplates systems and methods to support switching between computer audio and presenter audio during a conference call so that one and only one, not both, of the two audio streams is being transmitted to the rest of the conference at any time. More specifically, a switch engine is adopted, which separately takes one or more audio streams captured by a sound mixing component of the hosting device of a presenter and by a microphone attached to the hosting device as its two inputs, and outputs only one of the two inputs at any time. The selected audio output is then processed and transmitted to the rest of the conference with feedback from other audio sources canceled.

Abstract: This invention provides a method, system and apparatus for allowing media context sensitive SIP signaling exchange (such as voice) and call establishment while denying or challenging any other session description protocol (“SDP”) extension dialogs which might not be desired (such as instant messaging, video, Web broadcasting or pushing, data and/or application sharing and the like) by a user. The method and apparatus may further include defining user client media policy preferences, the user media policy preferences establishing the parameters for evaluating a media session request received by a user client, and providing the user client media policy preferences to a policy enforcement point device, the policy enforcement point device evaluating the media session request received by the user client and applying the user client media policy preferences to the media session request.

Abstract: In a computer system having a memory, a processor, and a network interface, a method for listening on multiple conferencing interfaces having the steps of loading a set of transport components into the memory; initializing each transport components of the set of transport components to listen on a particular conferencing interface using the network interface, each transport component of the set of transport components listening to a different conferencing interface; receiving an incoming call signal on the network interface having an incoming conferencing interface; processing the incoming call signal to detect the incoming conferencing interface; and launching an application based on the incoming conferencing interface.

Abstract: A method, an arrangement and a device for transmitting data via a plurality of channels. Additional data d are transmitted via at least one channel, and the transmission of the additional data d influences the transmission of the data in at least one of the further channels. The transmission of the data in the at least one of the further channels is influenced for example in such a way as to reduce interference, such as crosstalk, for instance, in the at least one of the further channels. The additional data d are advantageously transmitted during pauses in the user data transmission of the at least one channel.

Abstract: Briefly, in accordance with one or more embodiments, a method and device capable of canceling self-jammer and one or more other interfering signals in an radio frequency identification system or the like is disclosed.

Abstract: A crosstalk cancellation system and method is disclosed for use in a multi-channel communication system. Crosstalk which couples between channels is cancelled through use of an in-line FFE filter and in-line delay. A cross-connect system associated with each channel includes a cross-connect delay and cross-connect filter. The cross-connect system generates a cancellation signal for each of the channels, which is routed into a junction. The junction subtracts the cancellation signal from the received signal, which has also been delayed and filtered, to remove unwanted cross-talk. During training, cancellation magnitude is monitored at various delay offsets to determine which offset and corresponding filter coefficients, for each delay, maximizes cancellation. The filters are set with filter coefficients that maximize cancellation. The cross-connect delay with the maximum offset is set to zero and its calculated offset amount is established as the in-line delay offset.

Abstract: An apparatus for mixing audio signals in a voice-over-IP teleconferencing environment comprises a preprocessor, a mixing controller, and a mixing processor. The preprocessor is divided into a media parameter estimator and a media preprocessor. The media parameter estimator estimates signal parameters such as signal-to-noise ratios, energy levels, and voice activity (i.e., the presence or absence of voice in the signal), which are used to control how different channels are mixed. The media preprocessor employs signal processing algorithms such as silence suppression, automatic gain control, and noise reduction, so that the quality of the incoming voice streams is optimized. Based on a function of the estimated signal parameters, the mixing controller specifies a particular mixing strategy and the mixing processor mixes the preprocessed voice streams according the strategy provided by the controller.

Abstract: A method of controlling teleconference signals includes receiving, at a teleconference bridge, endpoint-generated audio signals from each of a plurality of participating endpoints arranged at a plurality of locations. At least two of the participating endpoints are acoustically collocated at one of the locations. A bridge-generated audio signal is generated for each of the participating endpoints based on a set of signals. The set of signals includes all of the endpoint-generated signals received at the teleconference bridge, exclusive of the endpoint-generated signals transmitted to the bridge from each of the acoustically collocated endpoints.

Abstract: A method for determining at least one operational transmit power Snk over a communication channel (Ln) coupled to a disturbance causing transceiver (n) for at least one tone (k), comprises the steps of provision of at least one reference victim communication channel (Lref) for representing at least one reference victim for all victim communication channels which are degraded by cross-talk interferences by said communication channel (Ln), and maximizing the data rate (Rref) over said at least one reference victim communication channel (Lref) under the constraint that the data rate over said communication channel (Ln) achieves a target rate (Rntarget) without exceeding a power constraint (Pnmax).

Abstract: A novel and useful mechanism for the detection and estimation of far-end cross talk (FEXT) caused by transmissions over adjacent wire pairs or cables. The detection and estimation technique of the invention can be used to identify FET impairments in Ethernet and DSL systems. The detection and estimation scheme is performed utilizing an adaptive FEXT canceller filter for detection purposes during normal operation of the system. FEXT filter tap coefficients are determined one at a time using a mechanism shared amongst all taps. Once adapted, the filter coefficients can be used to estimate FEXT strength and to determine spectral measurements of the FEXT impairment noise.

Abstract: A method of determining a channel estimate of a first transmission channel in a communications system. The method comprises deriving a first set of channel estimates from symbols received through said first transmission channel; deriving a second set of channel estimates from symbols received through a second transmission channel in the communications system; determining a scale factor between the first and second sets of channel estimates from a least squares error criterion; and determining the channel estimate of the first transmission channel as a channel estimate of the second transmission channel scaled by the determined scale factor.

Abstract: A video server adaptively selects an appropriate video encoding standard and an appropriate video encoding rate for encoding video content to be downloaded to one or more video client devices over a communication network. The video server adaptively selects the video encoding standard and the video encoding rate based upon factors such as the data transmission rate of the communication network, the processing speed of the video client device, or the type of the video content to be downloaded. Once the video encoding standard and the video encoding rate are selected, video content can be transcoded prior to downloading to one or more video client devices. During the download, the video content may be adaptively transrated, such as in response to a change in the data transmission rate of the communication network.

Abstract: A single chip integrated circuit wireless data processor demodulates N separate data signals from M separate antennas simultaneously. The multi-antenna processor can be coupled to a baseband processor on the IC, so that it responds to changing channel conditions between two access points, and selectively kicks in if there is noise, interference, frequency fading, a need for an enhanced data rate, a need for an increased operating range, etc. to improve a performance of the baseband processor.

Abstract: In one embodiment, the invention provides a method comprising loading a set of transport components into a memory; initializing each transport component of the set of transport components to listen on a particular conferencing interface using a listen string to associate with a network interface, each transport component of the set of transport components listening to a different conferencing interface, the listen string encapsulating parameters to one or more calls to listen; receiving an incoming call signal on the network interface having an incoming conferencing interface; processing the incoming call signal to detect the incoming conferencing interface; and launching an application based on the incoming conferencing interface. Other embodiments of the invention are also described.

Abstract: A method and apparatus are disclosed for canceling acoustic echo generated during transmission of text information in a communication system supporting a TTY/TDD service. When a mobile-to-land call is established between a mobile subscriber and a PSTN subscriber, a decoder notifies an encoder of detection of TTY/TDD text in a BS vocoder. The encoder then transmits a packet having silence information, which prevents acoustic echo reproduction of the text in the TTY/TDD device.

Abstract: A radio apparatus, to be used as a first ending node of a communication network, is connectable to a hands-free apparatus by a local radio link to form a hands-free system. A call made on the hands-free apparatus for an second ending node in the communication network is sent to the radio apparatus through a control channel on the local radio link. The radio apparatus and the hands-free apparatus arrange a voice channel to be established on the local radio link. The call is then sent to the second ending node. In a case where no voice channel is established within a predetermined time period after arranging the voice channel, the radio apparatus cancels the call to prevent a connection between the radio apparatus and the second ending node from being kept and charged.

Abstract: Systems and methods for reducing transmit echo in a transceiver are disclosed. A hybrid echo canceller includes a limited tap length FIR filter to cancel short-term echo, while an interpolated filter is used to cancel the long-tail echo. The limited tap length FIR filter adapts and calculates coefficients for each adapted tap. Taps of the interpolated filter, on the other hand, are adapted but coefficients are calculated for a subset of the taps. Various interpolation schemes may be applied to the calculated coefficients to associate a coefficient with each tap of the interpolated filter. The technique presented produces an effective filter length of N taps with a reduction in computation and signal processing resources. Preferred embodiments of the echo canceller may be construed as methods for reducing transmit echo.

Abstract: A device (2) and method (20) for providing a balanced parameterized description of a dynamical environment having a plurality of signal sources. The method (20) includes inverse transforming (25) an initial reduced cost set of discrete time parameters into a reduced cost set of continuous time parameters and then converting (26) the reduced cost set of continuous time parameters into a balanced converted set of parameters. A transforming step (26) for transforming the balanced converted set of parameters into a modified set of discrete time parameters is then effected and thereafter an estimating step (28) estimates from the modified set of discrete time parameters a balanced reduced cost set of discrete time parameters. Steps (25) to (28) are repeated until the balanced reduced cost set of discrete time parameters provides a rate of change value of a cost function that reaches a pre-determined value.

Abstract: A method and apparatus for noise cancellation in a multi-channel communication system is disclosed. In one embodiment this system is configured to cancel FEXT on a victim channel utilizing the signals received on the other channels. The processing benefits gained by a receiver's other filters, such as for example, the FFE and DFE filters, is utilized when generating a FEXT cancellation signal. As a result, the complexity of the apparatus that generates the FEXT cancellation signal may be made less complex since part of the processing burden is performed by other filter apparatus. In one configuration pre-code FEXT cancellation occurs in that a pre-code FEXT filter generates one or more pre-code FEXT cancellation signals corresponding to each of the other channels. The pre-code FEXT cancellation signals are combined, prior to transmission, with the signals associated with each of the other channels, to thereby pre-cancel FEXT prior to transmission.

Abstract: A transmit processor that includes a transmit frequency domain equalizer (TX FEQ) that pre-compensates packets in the frequency domain to flatten transmit filter response to improve spectral mask and reduce PER. The TX FEQ taps may be selected so that the average power at output equals the average power applied to the input. The TX FEQ may be designed to yield 64 QAM packets with no quantization distortion at the input of an Inverse Fast Fourier Transform (IFFT) processor. The 64 QAM constellation map points and TX FEQ gain values are selected to avoid exceeding the bit resolution of the IFFT processor. The tones may be bound together into zones to reduce implementation complexity by reducing the number of TX FEQ taps.

Abstract: The invention refers to magnetoelectric machines. It can be used when manufacturing different-purpose electric drives—for instance, in fans, compressors, electromobile's wheels, and so forth. It comprises a rotor made in the form of two disks, the teeth of which on the outer or inner circumference make up rotor poles and an axially magnetized cylindrical magnet (placed between said disks). The stator (made in the form of coils that are distributed over the circumference and that are installed predominantly in the space between the rotor poles) provides for the possibility of the end face interaction with the rotor poles. The rotor could be made as a multi-sectional unit. In this case said disks have the plate-like shape, owing to which fact the poles of one disk are located between the poles of another disk in one plane, while the magnets of adjacent section are oriented towards one another with like poles.

Abstract: A distributed packet-based audio conferencing system, method for packet-based audio conferencing, and a transceiver for use in such conferencing are disclosed. The system uses a collection of transceivers, with each conference participant connected to a local transceiver. When a participant speaks, the local transceiver is responsible for relaying the speaker's voice over a packet network by multicast transmission to transceivers local to each other conference participant. If multiple participants speak simultaneously, a multicast talk stream may originate from each speaker's local transceiver. The total number of simultaneous speakers, however, is limited by an arbitration function resident in each transceiver. The system reduces the costs associated with an always-up communication system.

Abstract: In a transceiver unit including an adaptive equalizer filter unit, apparatus is provided for reducing or compressing the number of bits representing an error signal. The apparatus replaces a plurality of the most significant logic signal bits with a single bits while transferring the sign logic signal bit and the logic signal bits of lesser significance unchanged. Because of the reduction in the number of logic signal bits, the number of components implementing the multiplier unit in the adaptive filter unit can be reduced (i.e., in each stage of the adaptive filter). The reduction of the apparatus implementing the processing the error signal results in the same equilibrium value of the error signal, however, the time to reach this equilibrium value is increased.

Abstract: Methods and apparatus are disclosed for dynamically discovering and utilizing an optimized network path through overlay routing for the transmission of data. A determination whether to use a default network path or to instead use an alternate data forwarding path through one or more overlay nodes is based on real-time measurement of costs associated with the alternative paths, in response to a user request for transmission of message data to a destination on the network. Cost metrics include delay, throughput, jitter, loss, and security. The system chooses the best path among the default forwarding path and the multiple alternate forwarding paths, and implements appropriate control actions to force data transmission along the chosen path. No modification of established network communication protocols is required.

Abstract: An axial gap electrical machine employs a coil assembly embedded in epoxy and coated with a rubbery substance having different degrees of hardness.Two magnet assemblies straddle a central coil assembly. The magnet assemblies each include a back iron casting located on the sides of the magnet assemblies away from the central coil assembly.

Abstract: A conference call sound level balancing method and system for selectively muting a conferee whose sound environment is loud enough to dominate other conferees of a conference call. The method is operable on a system that includes a network-based conference bridge and a sound equalization platform that is coupled to each telephone of a conference call. If one telephone connected to the conference call has a higher noise level than the other branches of the call, thus making it so the other branches cannot obtain speaking privileges, a predetermined touchtone key is entered at the noisy telephone. The sound equalization platform responds by automatically measuring the ambient sound levels of the respective branches of the conference call and balancing the speaking privileges for the respective call branches of the conference call.

Abstract: A software-implemented bridging routine is provided for a full duplex audio telephone system comprising units A, B and C. of which only unit A need have at least two lines and be full duplex. Unit A receives and speaker-reproduces an audio signal B+C that echoes with a time delay dt.sub.1 and is detected by unit A's microphone as a signal k.sub.b Bdt.sub.1 +k.sub.c Cdt.sub.1, where k.sub.b and k.sub.c are co-efficients. Unit A's host software determines time dt.sub.1 and generates a delay dt.sub.2 .apprxeq.dt.sub.1. Unit A's host processor sums its own microphone output with a signal generated by a non-destination unit that is delayed by time dt.sub.2. Thus, when communicating with unit B, unit A's software generates a host processor output signal A+k.sub.b Bdt.sub.1 +k.sub.c Cdt.sub.1 +Cdt.sub.2. The k.sub.b Bdt.sub.1 echo component is removed by unit A's AEC, which is coupled between units A and B. This causes unit B to receive the signal A+Cdt.sub.2 +k.sub.c Cdt.sub.1 from unit A. Since dt.sub.2 .apprxeq.

Abstract: A compact motor mount for cassette drive systems includes a base plate in the form of a laminate that includes at least one planar layer of ferromagnetic material sandwiched between two printed circuit boards. A capstan motor is supported by the base plate. The stator coil assembly of the capstan motor is mounted on one side of the base plate directly on one of the two printed circuit boards. At least one winding drive motor also is supported in common with the capstan motor by the base plate. The stator coil assembly of the drive motor is mounted on the other side of the base plate directly on the second of the two printed circuit boards. Each of the motors has a permanent magnet rotor mounted on a shaft and facing the respective stator coil assemblies across an air gap defined therebetween.