Abstract: Systems and methods of establishing a communications channel between a first telephony device and a second telephony device obtain information about the capabilities of various data network elements that can be used to establish the communications channel. The information about the elements is used to select a combination of elements that are used to establish the communications channel. A communications channel may also be monitored while it is in use. If the requirements for the channel change, the communications channel may also be changed accordingly. If any of the elements become incapable of providing the required level of service or functionality for a communications channel, the element may be removed from the communications channel.

Abstract: Encoding and decoding methods are provided. An information object is encoded according to a preset encoding scheme. A preset set of bits is selected from the encoded information object and the selected set of bits is output as a sample. A control packet including information on a sample number is generated. The sample is assembled with the control packet. The sample is modulated into a sound signal according to a preset scheme. An input sound signal is demodulated according to the preset scheme. Sample bits and control packet bits containing information of the sample are separated from each other in demodulated received bits. Address information corresponding to each bit of the sample is generated according to the control packet. A soft decision of the sample bits is summed for each piece of the generated address information. A sample decoding of the summed soft decision of the sample bits is performed.

Abstract: Disclosed is a signal processing apparatus that processes an input signal to accurately detect an abrupt change in the input signal in accordance with the degree of linear change of a phase component in a frequency domain. The signal processing apparatus includes a converter that converts the input signal into the phase component and an amplitude component in the frequency domain, a linearity calculator that calculates the linearity of the phase component in the frequency domain, and a determiner that determines presence of the abrupt change in the input signal based on the linearity calculated by the linearity calculator.

Abstract: Methods, devices and systems are provided for diagnosing a wireless module of a device. An exemplary method involves operating a second wireless module of the device to transmit data on a second wireless channel that overlaps, at least in part, a first wireless channel. A measured response associated with the first wireless channel concurrent to the data transmitted on the second wireless channel is obtained from the first wireless module and a remedial action is initiated with respect to the first wireless module based at least in part on the measured response.

Abstract: Communications equipment for use by a call center to enable communications between the call center and one or more user devices is provided, with the communications equipment comprising audio processing circuitry for processing audio signals received from the user devices over a communications network, wherein the processing circuitry is configured to perform acoustic echo cancellation and/or noise suppression and/or dereverberation processing on the received audio signals to cancel acoustic echoes and/or suppress noise and/or suppress reverberation that is present in the audio signals received from the user device.

Abstract: A packet voice transceiver adapted to reside at a first end of a communication network and to send and receive voice packets to and from a second packet voice transceiver residing at a second end of the communication network. The transceiver includes a voice encoder, a packet transmitter, a packet receiver, a voice decoder and a far-end echo canceller. The voice encoder receives an ingress audio signal and encodes the audio signal to produce an ingress packet voice signal. The packet transmitter transmits the ingress packet voice signal over the network to the second packet voice transceiver. The packet receiver receives an egress packet voice signal transmitted over the network by the second packet voice transceiver. The voice decoder decodes the egress packet voice signal to produce an egress audio signal. The far-end echo canceller reduces echo that is present in the egress audio signal and that originated at the second end.

Abstract: A method of activating voice control on a wireless device includes sampling signals from a plurality of sensors on the device, determining if the device is in a hands-on state by a user on the basis of the signal sampling, and enabling a voice activated detection (VAD) application on the device on the basis of the determination. A voice controlled apparatus in a wireless device includes a plurality of sensors arranged on the device, a microphone, a controller to sample signals from one or more of the plurality of sensors, a processor coupled to the controller, and a voice activated detection (VAD) application running on the processor coupled to the controller and the microphone.

Abstract: In order to reduce a delay at the time of detecting inputted optical signals in an optical receiver using a variable optical attenuator, the optical receiver includes: a variable optical attenuator that outputs optical signals by attenuating the intensity of inputted optical signals; a photoelectric converter that converts the optical signals into electric signals; an amplitude detection circuit that outputs an output voltage based on the amplitude of the electric signals; an optical attenuator control circuit that outputs signals for controlling an attenuation quantity of the variable optical attenuator based on the output voltage; a signal detection circuit that outputs signal detection output by comparing the output voltage and a signal detection threshold voltage, i.e., reference of signal detection, to each other; and a threshold control circuit, which monitors the output voltage, and which changes the signal detection threshold voltage when an output voltage change state becomes stable with time.

Abstract: A digital signal processor which performs digital signal processing of a digital signal includes a statistical analysis method which calculates a moving average and a standard deviation from the digital signal, performs statistical decision deciding whether or not the digital signal is within a predetermined range obtained from the moving average and the standard deviation, and corrects the digital signal outside the range within the range. Statistical analysis of the digital signal is performed, thereby suppressing transient changes without increasing the number of times of averaging during the digital signal processing.

Abstract: The present disclosure relates to an apparatus and a method for transmitting muting information in a wireless communication system, and to an apparatus and a method for acquiring channel state using same. In an exemplary embodiment, muting information includes: a first data field, having a serving cell for receiving from peripheral cells in a multi-cell environment, at least one of a CSI-RS pattern, the number of CSI-RS antenna ports, a CSI-RS duty cycle, and CSI-RS transmission subframe offset information, and using same for expressing the cycle and the offset of muting subframes, which pertain to information on a resource block that can generate interference between the peripheral cells and CSI-RS; and a second data field for expressing a specific muting pattern, which must be muted within the muting subframes, having either 12 bits or 28 bits that display muting application in a bitmap format.

Abstract: In embodiments, apparatuses, methods, and storage media may be described for reducing the overhead associated with the transmission of channel training signals from an eNodeB (eNB) of a wireless network. Specifically, the eNB may receive feedback from a user equipment (UE) regarding the received signal energy of a first and second beamformed signal produced with a first and second beamforming vector, respectively. The eNB may identify, based on the feedback of the received signal energy, a signal subspace and a null subspace. The eNB may then transmit a channel training signal to the signal subspace.

Abstract: A serial data transmitter utilizing switching equalization. The transmitter includes a first per-bit switcher configured to invert of every other bit of the stream of bits to form a switched signal; a filter configured to filter the switched signal to form a filtered signal; and a second per-bit switcher configured to invert every other bit of the filtered signal.

Abstract: One embodiment provides an apparatus. The apparatus includes a feedforward equalizer (FFE), an FFE data slicer and an FFE least mean square (LMS) module. The FFE data slicer is to threshold detect a sample of a received analog training signal, a result of the threshold detecting corresponding to an input signal digital decision. The FFE LMS module is to determine a plurality of FFE coefficients based, at least in part, on an output of the FFE data slicer and based, at least in part, on a mean square error corresponding to a difference between an equalized sample and a reference.

Abstract: A real-time sharing method, apparatus and system, which relate to the online application field to solve the problem that a user cannot actively send a request to another terminal by using a terminal according to a personal requirement to implement real-time sharing of resources of an online application of another terminal. The real-time sharing method for sharing of an online application includes: receiving a browsing request message sent by a first terminal, where the browsing request message carries an identifier of a second terminal; acquiring, according to the identifier of the second terminal, virtual machine information of a virtual machine that provides an online application service for the second terminal; and sending, to the first terminal, resource information sent to the second terminal by the virtual machine corresponding to the virtual machine information.

Abstract: An apparatus may include an antenna and a transceiver coupled to the antenna, the transceiver including a receiver operative to receive a radio-frequency (RF) signal and a transmitter operative to transmit a RF signal. The apparatus may also include an RF echo cancellation module coupled to the receiver and the transmitter, the RF echo cancellation module operative to generate an analog echo cancellation signal for the received RF signal based on a delayed version of the transmit RF signal. Other embodiments are disclosed and claimed.

Abstract: The present invention relates to methods, apparatuses and a computer program product for random access channel enhancement for carrier aggregation with different uplink/downlink configuration. The present invention includes detecting, at a user equipment, a downlink control channel in a specific downlink subframe on a first component carrier, receiving, at the user equipment, a downlink shared channel on the first component carrier, and transmitting, after a predetermined period, a random access channel in a specific uplink subframe on a second component carrier, the specific uplink subframe overlapping the specific downlink subframe on the first component carrier.

Abstract: In an example, time and frequency domain speech enhancement is implemented on a platform having a programmable device, such a PC or a smartphone running an OS. Echo cancellation is done first in time domain to cancel a dominant portion of the echo. Residual echo is cancelled jointly with noise reduction during a subsequent frequency domain stage. The time domain block uses a dual band, shorter length Adaptive Filter for faster convergence. Non-linear residual echo is cancelled based on an echo estimate and an error signal from the adaptive filters. A controller locates regions that had residual echo suppressed and which do not have speech and injects comfort noise. The controller can be full-duplex and operate non-linearly. An AGC selectively amplifies the frequency bins, based on the Gain function used by the residual echo and noise canceller.

Abstract: In some implementations, an apparatus includes an echo canceller that generates an echo interference compensation signal that compensates for an echo interference signal in a communication signal, a crosstalk canceller that generates a crosstalk interference compensation signal that compensates for a crosstalk interference signal in the communication signal, and a combiner that generates a combined interference compensation signal based on the echo interference compensation signal and the crosstalk interference compensation signal.

Abstract: Disclosed is a computer system comprising an audio processing module, an echo cancellation module and a control module. The audio processing module is configured to process an audio signal and output the processed audio signal to e.g. a speaker (or similar) of the system. The echo cancellation module is configured to cancel echo from an audio signal received via e.g. a microphone (or similar) of the system. The control module is configured to detect a condition indicative of nonlinear amplitude processing by the audio processing module and control said echo cancellation by the echo cancellation module based on the detected condition.

Abstract: An echo canceller for an IP network includes an adaptive filter that models the echo path and generates an estimate of the echo signal from a receiving input signal. The echo canceller subtracts the estimate of the echo signal from a sending input signal to generate a sending output signal with reduced echo. Variation in the echo delay is detected. A delay circuit compensates for the changes in the echo delay to provide proper time-alignment between the estimate of the echo signal and the sending input signal so that the echo signal will be more effectively cancelled.

Abstract: Signal transport with noise cancellation is contemplated. The noise cancellation may be facilitated with a transmitter configured to induce echoes in a signal desired for transport in order to facilitate subsequently retrieving signal components associated with noise influenced portions of the transported signal from non-noise influence portions of the transported signal.

Abstract: A system and method for identifying minor echoes present in an input signal in the situation where a set of major echoes has already been identified from the input signal. The method includes: computing a spectrum F corresponding to a sum of the major echoes; computing a weighted power spectrum SM of the spectrum F; subtracting the weighted power spectrum SM from a weighted power spectrum PIN of the input signal to obtain a difference spectrum; performing a stabilized division of the difference spectrum by a conjugate of the spectrum F to obtain an intermediate spectrum; computing an inverse transform of the intermediate spectrum to obtain a time-domain signal; and estimating parameters one or more of the minor echoes from the time-domain signal. The echo parameters are usable to remove at least a portion of the one or more estimated minor echoes from the input signal.

Abstract: A communication apparatus encodes a transmission target data according to a PAM5 mode to generate PAM signals, applies a pulse shaping to the generated PAM signals with a pulse shaping filter to provide pulse-shaped PAM signals, and inputs the pulse-shaped PAM signals into a digital-analog converter. The pulse shaping filter designates a transfer function G(z) as G1(z)=0.667+0.333z^?1 that is approximated to G0(z)=(?)+(?)z^?1.

Abstract: Application resource scheduler module is provided to achieve cooperative application workload scheduling for a consolidated virtual environment. The application resource scheduler aids an application workload scheduler that is part of a distributed computing application, such as Hadoop, to achieve a specified relative priority of the application workload virtual machines to other virtual machines in the virtual environment. The application resource scheduler assists in achieving cooperative workload scheduling by revising the amount of resources that the application workload scheduler sees as available and by setting resource controls for the virtual machines of the distributed computing application to influence the resources the virtual machines receive from the underlying consolidated virtual environment.

Abstract: In a conventional dereverberation system, when there is a fluctuating reverberation component, it has been difficult to determine, with high accuracy, a linear dereverberation filter for removing a non-fluctuating reverberation component. An algorithm integrating a dereverberation system using a linear filter and a dereverberation system using a non-linear filter includes the function of measuring the amount of fluctuation in transfer function in a latter-stage non-linear filter over time, and controls the strength of the non-linear filter over time based on the function. In this configuration, a strong non-linear process is implemented only when the fluctuation in transfer function is large, whereby distortion in speech components can be minimized.

Abstract: In a method for channel adaptation, an analog input signal is received with a bimodal receiver via a communications channel. The analog input signal is converted to a digital input signal with an analog-to-digital converter of a digital receiver of the bimodal receiver. Channel coefficients are detected for the digital input signal associated with the communications channel. The channel coefficients indicate a number of post-cursor taps of the bimodal receiver to be used to provide an equalized digital output signal from the digital input signal. It is determined whether the number of post-cursor taps or a value associated therewith is equal to or less than a threshold number. A switch from the receiving of the analog input signal by the digital receiver to an analog receiver of the bimodal receiver is made to provide the equalized digital output signal for the analog input signal.

Abstract: Apparatus and methods are described for space-efficient, high-speed data communications for integrated circuits. Bandwidth is multiplied by using multiple individual wireline communications channels coupled to form a communications lane. The data receiver for a channel implements symbol-rate equalization and crosstalk filtering that is space efficient, allowing high-speed data communications to be added as an ancillary function to an IC.

Abstract: A device (100) for processing a signal, wherein the device (100) comprises a processing unit (101) comprising an input terminal and two output terminals, wherein the processing unit is adapted for receiving a signal (104) at the input terminal, and wherein the processing unit is adapted for generating a reduction, signal based on the signal, wherein the reduction signal is adapted for reducing the signal. The device comprises further a first reproduction unit (102) coupled to the first output terminal of the processing unit, wherein the first reproduction unit is adapted for receiving and reproducing the signal, and a second reproduction unit (103) coupled to the second output terminal of the processing unit, wherein the second reproduction unit is adapted for receiving and reproducing the reduction signal.

Abstract: A method of removing a noise signal from an input signal, the method including receiving a pure noise signal and an input signal including the noise signal; determining whether the pure noise signal is a static noise signal or a non-static noise signal; and removing the noise signal from the input signal according to whether the noise signal is determined to be the static noise signal or the non-static noise signal.

Abstract: In an example embodiment, an example method is provided for echo mitigation in a conference call. In this method, a test audio signal is transmitted to a conference endpoint and as a result, an echo associated with the transmittal of the test audio signal is received. One or more parameters of the echo are then identified and an echo mitigation process is selected from multiple echo mitigation processes based on the identified parameters. The selected echo mitigation process is then applied.

Abstract: A method used in the receiver of a communication system is provided to suppress narrow band interferences, known as ingress, that are present in the communication channel, which is shared by a plurality of transmitters each having a channel pre-equalizer, by means of a Time Division Multiple Access (TDMA) scheme or Synchronous Code Division Multiple Access (SCDMA) scheme. The method comprises filtering the interferences using Infinite Impulse Response (IIR) notch filters, and providing IIR all-pass filters devised to compensate for substantially all phase distortions introduced by the notch filters. The method also includes an algorithm to detect the narrow band interferences and an adaptive equalizer to equalize the channel by computing the coefficients of the transmitters' pre-equalizers, and arranged such that the equalization of the channel is isolated from the suppression of the interferences.

Abstract: For VoIP calls over a communication system in which a remote terminal is requesting to a hub-gateway, bandwidth may be allocated according to a dynamic technique based on the packet rate between an access device and the remote terminal by sending a bandwidth request message from the access device to the hub-gateway or to the remote terminal containing information on bandwidth allocation requirements for a given set of VoIP calls. The hub-gateway or the remote terminal identifies the received messaged information and analyzes the information on bandwidth allocation requirements contained in the messaged information. In addition, the hub-gateway, or the remote terminal, upon necessity, sends a request to the hub-gateway for bandwidth to be allocated in accordance with the analyzed information on bandwidth allocation requirements.

Abstract: Cross-talk reduction and/or cancellation systems and methods are provided herein. In exemplary embodiments, a far-end acoustic signal is delayed by M samples. Additionally, a cross-talk estimate value for the delayed far-end acoustic signal may be subtracted from an input acoustic signal. The cross-talk estimate value is a scaled version of filter outputs generated by a finite impulse response filter that utilizes predetermined filter coefficients. The filter outputs are scaled using a dynamic gain value.

Abstract: A channel in a multiple carrier communication system is equalized by computing a desired spectral response, shortening the impulse response of the channel so that a significant part of an energy of the impulse response is confined to a region that is shorter than a target length and filtering the signal based on the desired spectral response. A multiple carrier communication system may include a primary impulse shortening filter that receives an output signal of an analog to digital converter and accepts coefficients. A secondary impulse shortening filter may receive the output signal of the analog to digital converter, output an output signal, and pass coefficients to the primary impulse shortening filter. A reference signal generator may output a reference signal. A comparator may compare the output signal and the reference signal and output a resulting error signal. An adaptive processor may compute coefficients for the secondary impulse shortening filter based on the error signal.

Abstract: A method and apparatus for suppressing silence frames in a stream of media includes receiving a stream of media from a user and suppressing at least one silence frame from the received stream of media. The suppressed silence frame includes an initial silence frame situated before a first media frame and/or a silence frame situated between two successive media frames.

Abstract: A system and method for identifying minor echoes present in an input signal in the situation where a set of major echoes has already been identified from the input signal. The method includes: computing a spectrum F corresponding to a sum of the major echoes; computing a weighted power spectrum SM of the spectrum F; subtracting the weighted power spectrum SM from a weighted power spectrum PIN of the input signal to obtain a difference spectrum; performing a stabilized division of the difference spectrum by a conjugate of the spectrum F to obtain an intermediate spectrum; computing an inverse transform of the intermediate spectrum to obtain a time-domain signal; and estimating parameters one or more of the minor echoes from the time-domain signal. The echo parameters are usable to remove at least a portion of the one or more estimated minor echoes from the input signal.

Abstract: The present invention provides a processing method based on an OFDM-TDMA two-way service and a communications device. The method includes assigning timeslots in a TDMA frame structure to multiple users in a TDMA manner, so that each user has a different timeslot, and each timeslot of the users is simultaneously used for uplink and downlink communication. The methods further includes simultaneously sending a local-end OFDM signal and receiving a remote-end OFDM signal in each timeslot of the users, and cancelling, in an OSD manner, interference caused by an echo OFDM signal, which is produced according to the local-end OFDM signal, to the remote-end OFDM signal, where a spectrum of the echo OFDM signal and a spectrum of the remote-end OFDM signal overlap.

Abstract: Methods and systems for a subband acoustic echo canceller with integrated voice activity detection are disclosed and may include adjusting transmit and/or receive powers of wirelessly communicated audio signals based on voice activity detection via subband analysis of the wirelessly communicated audio signals. The receive power may be adjusted by utilizing a reduced duty cycle, or by conveying voice activity detection information via an asynchronous control channel in a Bluetooth application. A plurality of subbands may be generated utilizing a fast Fourier transform, and a first subset of the subbands corresponding to voice activity may be selected and a second subset of the subbands may be selected that corresponds to background noise. The processing of the subsets may be dynamically adjusted due to variations in the voice activity or background noise. Comfort noise may be generated and transmitted at a reduced bandwidth utilizing the second subset of the subbands.

Abstract: A system includes a full-duplex driver to drive signals on a load. A hybrid element connected with the full-duplex driver controls a flow of transmission and receipt of the signals. A gain control element connected with the full-duplex driver tunes a transconductance of the full-duplex driver to match an impedance of the load. The controlled gain is based on a leakage voltage value of the full-duplex driver.

Abstract: In full duplex operation, a first signal is transmitted from a wireless device while a second signal in the same frequency band as the first signal is received at the device. A third signal can be generated that accounts for channel characteristics on the transmitted first signal. The third signal can be applied to the received second signal to compensate for interference caused by the transmitted first signal.

Abstract: An impedance adjustment method for a communication device, wherein the communication device has a plurality of impedance paths for selection, includes: selecting an initial impedance path; and utilizing a predetermined algorithm to examine a portion of the plurality of impedance paths by starting from the initial impedance path for selecting an optimized impedance path for the communication device. A delay capacitance adjustment method for a communication device, wherein the communication device has a plurality of delay capacitance paths for selection, includes: selecting an initial delay capacitance path; and utilizing a predetermined algorithm to examine a portion of the plurality of delay capacitance paths by starting from the initial delay capacitance path for selecting an optimized delay capacitance path of the communication device.

Abstract: A method for reducing interference in a received downhole telemetry signal includes: segmenting a received signal; windowing each signal segment; transforming each windowed signal segment into a complex variable domain to generate a plurality of complex variable domain segments with an in-phase component vector I and a quadrature component vector Q; calculating a real amplitude vector A from the I and the Q vectors; filtering interferers in the amplitude vector A for each complex variable domain segment to generate a filtered amplitude vector Ã; recalculating an amplitude of the amplitude vector A using the filtered amplitude vector à to generate an output amplitude vector Â; scaling the I and the Q vectors by a factor Â/A to generate an output in-phase component vector I? and an output quadrature component vector Q?; and transforming I? and Q? into the time domain to provide an interference-reduced output signal in the time domain.

Abstract: Methods and devices for improved echo cancellation in high noise environments, such as in-car speakerphones and the like, are provided that improve cancellation convergence in a high noise environment and in the presence of double talk. A new update method enables a robust echo canceller to update with reduced concern for the canceller being in a receive state or a double talk state. If a signal generated by a local talker is substantially the same as high noise, in terms of corruption of the echo canceller, the echo canceller converges in a robust manner even with continuous high noise at the input.

Abstract: In an example embodiment, a control connection is introduced between an adaptive jitter buffer (AJB) and an adaptive bulk delay (ABD) buffer of an echo canceller (ECAN) in an Internet Protocol (IP) conference bridge. The control connection allows the AJB to control the amount of delay inserted by the ABD in the ECAN convolution processor (CP) signal path. The adjustment in ABD delay restores the time alignment of the ECAN internal echo estimate and offsets variations in echo tail delay introduced by network induced AJB delay changes. Time-invariance is preserved in the echo tail path.

Abstract: Interface circuitry and method for transmitting and receiving downstream and upstream data signals simultaneously via a common conductor pair. The composite signal containing the downstream and upstream data signal components being conveyed by the common conductor pair is isolated, e.g., via signal filtering or buffering, and combined with an appropriately scaled inverse replica of the outgoing upstream data signal to subtract out upstream data signal components and thereby provide the downstream data signal substantially free of any upstream data signal components.

Abstract: Embodiments are directed to eliminating false audio using an egress gateway in a communications network. At least one false audio packet is received by an egress gateway. The false audio packet includes false audio. A DTMF packet is received by the egress gateway. The DTMF packet is received subsequent to the at least one false audio packet. The false audio in the false audio packet is replaced with a substitute signal by the egress gateway.

Abstract: The present disclosure relates to an apparatus and a method for transmitting muting information in a wireless communication system, and to an apparatus and a method for acquiring channel state using same. In an exemplary embodiment, muting information includes: a first data field, having a serving cell for receiving from peripheral cells in a multi-cell environment, at least one of a CSI-RS pattern, the number of CSI-RS antenna ports, a CSI-RS duty cycle, and CSI-RS transmission subframe offset information, and using same for expressing the cycle and the offset of muting subframes, which pertain to information on a resource block that can generate interference between the peripheral cells and CSI-RS; and a second data field for expressing a specific muting pattern, which must be muted within the muting subframes, having either 12 bits or 28 bits that display muting application in a bitmap format.

Abstract: System and methods provide acoustic echo monitoring and cancellation for real time media processing in an internet protocol (IP) media server in an IP network. An echo monitor is configured to selectively compare audio streams into and out of the IP media server through a selected port. The comparison determines an occurrence of an echo. An echo canceller in communication with the echo monitor is configured to respond to the determination by the echo monitor so as to remove the echo from at least one of the audio streams. A talk burst detector may be used to detect speech in at least one of the audio streams through the selected port. The echo monitor selectively compares the audio streams in response to a signal from the talk burst detector that indicates detection of speech.

Abstract: Provided is a private branch exchange capable of independently adjusting call voice level and call quality according to a connection type of a communication line or a telephone.

Abstract: A method for computing a gain control metric used in controlling gain in a wireless repeater operates to store correlation and normalization values associated with the gain control metric for the previous N samples in registers. For each new sample of the gain control input signal, the correlation and normalization values are computed by discarding the multiplication terms of the obsolete sample and adding the multiplication terms of the new sample to the stored correlation and normalization values. In this manner, the complexity of the computation is greatly reduced and the complexity of the computation does not increase with the integration length.