Abstract: Methods, systems, and apparatus, including computer programs encoded on a computer storage medium, for reducing audio noise are disclosed. In one aspect, a method includes the actions of receiving first audio data of a user utterance. The actions further include determining an energy level of second audio data being outputted by the loudspeaker. The actions further include selecting a model from among (i) a first model that is trained using first audio data samples that each encode speech from one speaker and (ii) a second model that is trained using second audio data samples that each encode speech from either one speaker or two speakers. The actions further include providing the first audio data as an input to the selected model. The actions further include receiving processed first audio data. The actions further include outputting the processed first audio data.

Abstract: A signal transceiver circuit, a method of operating a signal transmitting circuit, and a method of setting a delay circuit are provided. The signal transceiver circuit is used to send an output signal and receive an input signal, and includes: a delay circuit for delaying a first clock to generate a second clock; a first digital-to-analog converter (DAC) for converting a first digital signal into the output signal according to the first clock; a second DAC for converting the first digital signal into an echo cancellation signal according to the second clock; an analog front-end circuit for receiving the input signal and the echo cancellation signal and generating an analog signal; and an analog-to-digital converter (ADC) for converting the analog signal into a second digital signal.

Abstract: A system for interference mitigation includes: a first transmit coupler; a receive-band noise cancellation system; a first transmit-band filter; a second transmit coupler; a first receive coupler; a transmit-band noise cancellation system; a first receive-band filter; and a second receive coupler.

Abstract: A mobile communication device that comprises a microphone, a speaker, an input output port and one or more modules; wherein the microphone is configured to receive, during a conference call, (a) an acoustic signal from a first person, and (b) a cross echo from a speaker of another mobile communication device that participates in the conference call, the cross echo is generated as a result of the acoustic signal from the first person; and wherein the one or more modules are configured to cancel the cross echo based, at least in part, of the acoustic signal of the first person.

Abstract: A signal processor for performing signal enhancement, the signal processor comprising: an input-terminal, configured to receive an input-signaling; an output-terminal; an interference-cancellation-block configured to receive the input-signaling and to provide an interference-estimate-signaling and an interference-cancelled-signal based on the input-signaling. The signal processor further comprises a feature-block configured to provide a combination-feature-signal based on the interference-cancelled-signal and the interference-estimate-signaling; and a neural-network-block configured to apply model parameters to the combination-feature-signal to provide a neural-network-output-signal to the output-terminal.

Abstract: A system for reducing near-end crosstalk (NEXT) includes a first electrical component contributing to a first NEXT pulse of a first polarity and a second electrical component receiving the first NEXT pulse at a first time delay (td). A circuit couples the first electrical component to the second electrical component to create a second NEXT pulse of a second polarity and delays the second NEXT pulse by an amount equal to the first time delay (td). The first and second NEXT pulses are combined to reduce the first NEXT pulse. A method for reducing NEXT includes coupling a first component contributing to a first NEXT pulse of a first polarity and a second component to create a second NEXT pulse of a second polarity, delaying the second NEXT pulse, and combining the second NEXT pulse with the first NEXT pulse to reduce the first NEXT pulse.

Abstract: An echo cancellation circuit is coupled to a receiving circuit and a transmitting circuit of an electronic device, and the transmitting circuit includes an output transistor. The echo cancellation circuit includes first and second transistors, first and second resistor-capacitor networks (RC networks), and first and second resistors. The first transistor has a first gate, a first drain and a first source. The second transistor has a second gate, a second drain and a second source. The first drain and the second drain are coupled to the receiving circuit. The first RC network is coupled between the gate of the output transistor and the first gate. The second RC network is coupled between the first gate and the second gate. The first resistor is coupled between the first source and a reference voltage. The second resistor is coupled between the second source and the reference voltage.

Abstract: Provided are an audio processing device, an audio processing method, an information processing device, and a computer program that perform echo cancellation corresponding to double talk. The audio processing device includes an estimation unit that estimates a filter representing a transmission characteristic from a speaker where a reference signal is output to a microphone in which the reference signal sneaks, an adjustment unit that adjusts a step size on the basis of a filter update coefficient estimated by the estimation unit, and an update unit that updates the filter according to the update coefficient and the step size. The adjustment unit adjusts the step size on the basis of a ratio of power of the filter update coefficient to maximum power of the filter.

Abstract: A system configured to perform cascade echo cancellation processing to improve a performance when reference signals are asymmetric (e.g., dominant reference signal(s) overshadow weak reference signal(s)). The system may perform cascade echo cancellation processing to separately adapt filter coefficients between the dominant reference signal(s) and the weak reference signal(s). For example, the system may use a dominant reference signal to process a microphone audio signal and generate a residual audio signal, using the residual audio signal to adapt first filter coefficient values corresponding to the dominant reference signal. Separately, the system may use a weak reference signal to process the residual audio signal and generate an output audio signal, using the output audio signal to adapt second filter coefficient values corresponding to the weak reference signal.

Abstract: A method includes receiving a video data associated with a user in an electronic conference. The method further includes receiving an audio data associated with the user in the electronic conference. It is appreciated that the video data is processed to determine one or more actions taken by the user, and wherein the processing identifies a physical surrounding of the user. The method further includes identifying a portion of the audio data to be suppressed based on the one or more actions taken by the user during the electronic conference and further based on the identification of the physical surrounding of the user.

Abstract: An interference canceling subsystem for a bidirectional communications network includes an input interface configured to receive a first data signal from a first transceiver of the network, an output portion configured to receive a second data signal from a second transceiver of the network, a first signal path connecting the input interface to the output portion, a second signal path connecting the output portion to the input interface, and a first interference canceler disposed between the output portion and the input interface along the second signal path. The first signal path is configured to relay the first data signal from the input interface to the output portion. The interference canceler is configured to (i) relay the second data signal from the output portion to the input interface, and (ii) remove portions of the first data signal from the relayed second data signal prior to reaching the input interface.

Abstract: Methods, circuitries, and systems for transmitting data upstream between a first device and a second device and downstream between the second device and the first device are disclosed. A method includes determining an upstream crosstalk effect for an upstream channel and determining a downstream crosstalk effect for a downstream channel. The crosstalk effect includes near end crosstalk from a third device that is non-co-located with respect to the first device and the second device. A data rate of an upstream transmission or downstream transmission is adjusted based on the upstream crosstalk effect and the downstream crosstalk effect.

Abstract: A method is provided for accessing a memory via at least one address, wherein the at least one address comprises a codeword of a code. Corresponding devices are also described.

Abstract: A method and apparatus of transmitting data from a transmitter device to one or more receiver devices, each of the one or more receiver devices being connected to the transmitter device via a respective wire connection, the transmitter device being operable to transmit signals onto the wire connections over one or more different channels, the method comprising: iteratively performing steps (i) to (iii) until one or more criteria are satisfied, thereby determining current transmission powers for transmitting data, wherein: step (i) comprises determining values of a first function depending on one or more channel properties; step (ii) comprises, using the determined value of the first function, determining transmission powers for transmitting data; and step (iii) comprises determining whether or not the determined transmission powers satisfy one or more criteria; and thereafter transmitting data at the current transmission powers.

Abstract: An audio system, including an echo canceler being configured to receive a first reference signal and a microphone signal, and to minimize an echo signal of the microphone signal, according to the first reference signal, the echo signal being a component of the microphone signal correlated to the first reference signal, to produce a residual signal; a post filter configured to receive a second reference signal and the residual signal, and to suppress at least one residual component correlated to the second reference signal, according to the second reference signal, to produce an estimated voice signal, wherein the first reference signal is received from a first location of an audio processing chain and the second reference signal is received from a second location of the audio processing chain, the first location and the second location being separated by at least one audio processing module of the audio processing chain.

Abstract: This application discloses a decoding method for an OvXDM system, including: generating an augmented matrix B related to a received symbol information sequence; performing singular decomposition on the augmented matrix B; and performing decoding by using a total least square method, to obtain a decoded output information sequence. This application further discloses an OvXDM system. In a specific implementation of this application, decoding is performed by using the total least square method.

Abstract: An interlocking circuit and procedure for suppressing an echo on a receiving line are provided. The interlocking circuit blocks the receiving line when there is a signal on a transmission line. The receiving line has a transmission side at which the echo signal is located and a receiving side at which the echo signal can be blocked. The echo signal of a CAN driver is suppressed such that an actuating component for a headlamp light matrix does not receive any unwanted commands.

Abstract: The invention relates to a computer-implemented method for updating at least one frequency-domain filter coefficient Wi,j(k) of an echo canceller having at least one channel and at least one segment per channel, the filter coefficients of the echo canceller being updateable in the frequency domain at a time block m comprising: determining a canceller output Em(k) over the mth time block; determining a canceller error ?(k) over the m?th time block; determining a look-backward error (k); determining a look-forward error {right arrow over (?)}(k) as {right arrow over (?)}(k)=Em(k)??{right arrow over (?)}(k) wherein {right arrow over (?)}(k) is based on ? i = 0 C - 1 ? ? j = 0 S - 1 ? ? ? ( k ) ? X i , j , m - ? ? * ? ( k ) ? a = 0 C - 1 ? ? b = 0 S - 1 ? ? X a , b , m - ? ? ( k ) ? 2 + ? ? X i , j , m ? ( k ) ; determining an optimal update step-size ?i,j,m(k) from said canceller output Em(k) over the mth time blo

Abstract: Synchronizing methods and architectures for cable modems to transmit and receive Full Duplex (FDX) resource block allocations (RBAs) using filter switching and coordinated updating of equalization coefficients. A cable modem including a block of switchable filters, an analog front end (AFE) and a PHY/MAC System on a Chip (SoC) tuner to, at least in part, provide signals to switch the switchable filters in accordance with the RBA changes and synchronize updating tuner equalizations to match filter switching in a coordinated manner by marking received data at the AFE.

Abstract: Systems and methods are provided for full duplex DOCSIS cable modem echo cancellation with training. During reception of downstream signals, echo effects on downstream signals may be determined, with at least some of the echo effects corresponding to concurrently transmitted upstream signals. Echo cancellation corrections may be determined based on the determined echo effects, and the echo cancellation corrections may be applied during processing of the downstream signals. The echo cancellation corrections may include one or both of ACI (adjacent channel interference) cancellation corrections and ALI (adjacent leakage interference) cancellation correction. The echo cancellation may include or be based on preforming echo cancellation training, during active communication and based on one or both of the downstream signals and the upstream signals.

Abstract: A device that determines an echo latency estimate by subsampling reference audio data. The device may determine the echo latency corresponding to an amount of time between sending reference audio data to loudspeaker(s) and microphone audio data corresponding to the reference audio data being received. The device may generate subsampled reference audio data by selecting only portions of the reference audio data that have a magnitude above a desired percentile. For example, the device may compare a magnitude of an individual reference audio sample to a percentile estimate value and sample only the reference audio samples that exceed the percentile estimate value. The device generate cross-correlation data between the subsampled reference audio data and the microphone audio data and may estimate the echo latency based on an earliest significant peak represented in the cross-correlation data.

Abstract: Computer-implemented methods and systems are disclosed for optimizing the performance of wireless networks by automatically capturing wireless traffic and other radio frequency (RF) signal data in the network and analyzing the data to identify network anomalies and to determine one or more solutions, without human intervention.

Abstract: A system to implement a communication line coding scheme using a non-complex bit-to-symbol mapping, a forward error correction (FEC) coding, and an additive bit scrambler after the FEC at the PHY layer is provided. The system may be a part of or implemented by an automobile component. The system may be a PHY device configured to convert data from the MAC layer into 2D-PAM3 symbols that are transmitted across a communication link at a predetermined transmission rate, such as to be compliant with a communication standard. The PHY device may select characteristics of the conversion, such as the FEC coded symbol, based on the target transmission rate. The PHY device may include a transceiver, and may convert the data from MAC layer to PHY layer and back.

Abstract: An echo time delay detection method, an echo elimination device, and terminal equipment are provided. The echo time delay detection method includes: obtaining an audio signal transmitted through air and acquired by an audio input device of terminal equipment when detecting a call access request, the audio signal corresponding to an audio output by an audio output device of the terminal equipment; converting the audio signal to obtain a signal to be processed; determining a reference signal corresponding to the audio signal; and determining an echo time delay value according to the reference signal and the signal to be processed.

Abstract: A method for processing audio data includes obtaining a noise frame of an audio signal, and decomposing the current noise frame into a noise low-band signal and a noise high-band signal; and encoding and transmitting the noise low-band signal by using a first discontinuous transmission mechanism, and encoding and transmitting the noise high-band signal by using a second discontinuous transmission mechanism. According to the present disclosure, different processing manners are used for the high-band signal and the low-band signal, calculation loads and encoded bits may be saved under a premise of not lowering subjective quality of a codec, and bits that are saved may help to achieve an objective of reducing a transmission bandwidth or improving overall encoding quality.

Abstract: Embodiments are directed to a communication-session (CS) module that monitors the status of a communication session and at least partially controls the status. The CS module provides a party participating in a communication session an indication of whether a status of a microphone the user is employing to participate in an audio and/or video communication session is in a mute (or off) state, or whether the microphone status is in a live (or on) state. Likewise, a CS module may indicate to a user whether a status of a camera that the user is employing to participate in a video communication session is in an off state, or whether the camera status is in an on (or live) state. As such, a CS module provides users reminders of whether or not the microphone and/or camera, is in a live state, a mute state, or is otherwise turned on or off.

Abstract: A method and apparatus for cancelling self-interference in a cable network implementing full duplex (FDX). A processor in a cable modem may obtain a frequency response of an echo channel for at least one sub-channel in an FDX band and calculate filter coefficients based on the frequency response, and an interference canceller cancels the self-interference in the downstream signal. The interference canceller includes filters in a digital domain to generate an interference cancellation signal from a reference signal. In another example, the processor may obtain frequency responses of an echo channel for all sub-channels in a FDX band, combine the frequency responses for the FDX band, and generate a channel impulse response for the FDX band from the combined frequency response. The interference canceller may then cancel the self-interference in the downstream signal in an analog domain based on the channel impulse response.

Abstract: An apparatus, such as a base station or a user equipment, includes a transceiver configured to receive a first signal that is a superposition of symbols transmitted concurrently by users in shared resources of an air interface. The apparatus also includes a processor configured to iteratively cancel, for the users, interference produced by the symbols transmitted by other users on the basis of log likelihood ratios (LLRs) that represent likelihoods that previous estimates of the symbols transmitted by the other users are correct. The processor is also configured to iteratively decode the symbols transmitted by the users after canceling the interference produced by the symbols transmitted by the other users.

Abstract: Step size can be used to slow or freeze the adaptive filter to improve AEC system performance, such as during double talk events. An AEC control system may be used to adjust the step size based on an echo-to-disturbance energy ratio (EDER). The algorithm adjusts the step size to lower the adaptation rate when the EDER is small (or the combination of the near signal and noise is large compared to the echo) and raise the adaptation rate when the EDER is large (echo is large compared to a combination of near signal and noise).

Abstract: The invention relates to a peer-to-peer network having at least one first node with a first clock module and part of a peer-to-peer application. Also included is at least one second node with a second clock module and part of the peer-to-peer application. At least one communication connection between the first node and the second node is establishable. The first node comprises at least one first synchronization clock module. The second node comprises at least one second synchronization clock module. At least the first synchronization clock module is configured to transmit at least one first synchronization clock message to the second synchronization clock module via the communication connection, the second synchronization clock module is configured to synchronize the clock signal of the second clock module to the clock signal of the first clock module based on synchronization information included in the first synchronization clock message.

Abstract: An originated packet is to be forward-propagated along a path to a destination device, with the path including one or more intermediary node devices. A path quality indication representing a reported error having been detected by at least one of the intermediary node devices residing along the path, is incorporated in the forward-propagating packet. The destination device incorporates the path quality indication in an acknowledgement packet directed to the originator of the packet.

Abstract: A system configured to improve speech quality by performing residual echo suppression (RES). The system may detect when double talk is present in a voice conversation and may use different attenuation parameters based on a frequency of audio data. The system may perform RES on the audio data using a low attenuation value for low frequencies and a high attenuation value for high frequencies and determine that double talk is present when a difference in energy level between the low frequencies and the high frequencies is below a threshold. If double talk is present, the RES may generate output audio data using a low attenuation value for low frequencies and a high attenuation value for high frequencies. If double talk is not present, the RES may generate output audio data using a high attenuation value for low frequencies and a high attenuation value for high frequencies.

Abstract: Disclosed herein is a multi-stage echo cancellation scheme. The disclosed embodiments include an apparatus and method for monitoring and canceling echoes greater than 25 dB in a coaxial cable plant. The method includes obtaining echo channel estimate coefficients from a cable node. The method determines a location and strength of each partial echo in an impulse response using the echo channel estimate coefficients. Optionally, the method determines a frequency response of each partial echo in the impulse response.

Abstract: An echo time delay detection method, an echo elimination device, and terminal equipment are provided. The echo time delay detection method includes: obtaining an audio signal transmitted through air and acquired by an audio input device of terminal equipment when detecting a call access request, the audio signal corresponding to an audio output by an audio output device of the terminal equipment; converting the audio signal to obtain a signal to be processed; determining a reference signal corresponding to the audio signal; and determining an echo time delay value according to the reference signal and the signal to be processed.

Abstract: A method of processing an audio signal comprising a plurality of audio samples, the method comprising: receiving an audio sample of the plurality of audio samples; on determining that a peak amplitude of the audio sample falls outside a threshold range: suppressing the audio sample to within the threshold range by applying a strictly increasing non-linear function to the audio sample; and outputting the suppressed audio sample; and on determining that the peak amplitude of the audio sample falls within the threshold range or is equal to an upper or lower limit of the threshold range: outputting the received audio sample.

Abstract: An apparatus for generating a modified speech signal from a speech input signal which has a plurality of speech subband signals, the modified speech signal having a plurality of modified subband signals is provided, having: a weighting information generator for generating weighting information for each speech subband signal depending on a signal power of said speech subband signal, and a signal modifier for modifying each speech subband signal by applying the weighting information on said speech subband signal to obtain a modified subband signal.

Abstract: A radar system including a narrowband radar receiver configured to receive pulsed radar return signals and a wideband receiver configured for receiving wideband signals. A noise data processor is configured to identify impulse noises by analyzing wideband signals received by the wideband receiver and a radar processor is configured to cancel the identified impulse noises from pulse signals received by the radar receiver. The wideband impulse noises utilized to cancel noise from the pulse signals corresponds to the same time period sweep of detection as that of the pulse signals.

Abstract: A method and system for determining short-timescale traffic rates from time stamped packet data is described. The method and system provides for the calculation of traffic rates, or bandwidths, in bits-per-second or packets-per-second over short time windows, using time-stamped packet records as input.

Abstract: A gain control system for applying gain to a far-end signal, the system comprising: a signal identifier configured to detect an echo of the far-end signal in a microphone signal; and a path estimator configured to estimate a characteristic of an echo path of the detected echo, wherein: the signal identifier is further configured to detect a near-end signal from the microphone signal; and in response to detecting the near-end signal, the gain control system is configured to adjust the gain applied to the far-end signal in dependence on the estimated characteristic of the echo path.

Abstract: An approach is disclosed for synchronization of frequency and/or phase in a full duplex network. The method may comprise sending a timing beacon over a single full duplex radio channel from a timing master to a timing slave; receiving the timing beacon over the single full duplex radio channel at the timing slave; sending a retransmitted beacon over the single full duplex radio channel from the timing slave to the timing master; and sending a delta of the timing beacon and the retransmitted beacon from the timing master to the timing slave. The beacon may contain a timestamp. The retransmitted beacon may include a delay calculation. A hybrid analog/digital self-interference cancellation system may be used at the timing master or the timing slave. The method may also include simultaneously sending and receiving the synchronization information over the single full duplex radio channel.

Abstract: A method and apparatus for achieving simultaneous transmit and receive operation with digital cancelling based upon probe waveforms is described. Digital cancelling based upon probe waveforms enables adjacent transmitting and receiving channels to transmit and receive correlated signals.

Abstract: A multimedia system includes a source device for providing a media stream and a sink device for playing the media stream. The source device encapsulates the media stream into data packets with corresponding timestamps associated with a first wall time, and transmits the data packets to the sink device based on the timestamps and the first wall time. The sink device provides a second wall time based on the second global clock, synchronizes the second wall time with the first wall time through a network protocol, generates a local media clock, and locks the local media clock to the second global clock. The sink device decapsulates the data packets, and then plays the media stream with the locked local media clock. A play-out time of each data packet is determined by a corresponding timestamp and the second wall time.

Abstract: An echo-cancelling device includes an audio input/output (I/O) terminal, an audio-receiving module, an analog-to-digital (A/D) converter, and a processor is provided. The A/D converter is electrically connected to the audio-receiving module. The processor is electrically connected to the A/D converter and the audio I/O terminal. The audio I/O terminal receives an external reference signal from an electronic device. The audio-receiving module receives an input audio and an output audio from the electronic device, and generates an analog input signal having the input audio and the output audio. The A/D converter generates a digital input signal according to the analog input signal. The processor cancels the output audio to generate a second (digital) sound signal according to the digital input signal and the external reference signal. Finally, the processor transmits the second (digital) sound signal to the electronic device via the audio I/O terminal.

Abstract: A method for determining a distance between a ranging device and target, wherein a valid interference profile, if already available, is subtracted from a frequency-domain echo profile that is analyzed to determine the distance of the target, where to obtain or update the valid interference profile, a low-frequency portion of the frequency-domain echo profile is stored as a temporary interference profile after each measurement, and if the target is not proximal to the ranging device and the valid interference profile is not available, the temporary interference profile is saved as the valid interference profile, otherwise if the valid interference profile is available, the valid interference profile is updated via the temporary interference profile, otherwise if the target is distal and the valid interference profile is not yet available, the temporary interference profile is first subjected to a quality check before being saved as the valid interference profile.

Abstract: A communication apparatus creates a network, assigns information necessary for performing communication in the created network to each of other communication apparatuses which have joined the network, and searches for a communication apparatus having a specific service function. Upon completion of assignment of the information necessary for performing communication in the created network, the communication apparatus starts the search.

Abstract: Systems and methods for disabling adaptive echo cancellation functionality for a temporal window are provided herein. In some embodiments, audio data may be received by a voice activated electronic device, where the audio data may include an utterance of a wakeword that may be subsequently followed by additional speech. A start time of when the wakeword began to be uttered may be determined by the voice activated electronic device, and the voice activated electronic device may also send the audio data to a backend system. Adaptive echo cancellation functionality may be disabled at the start time. The backend system may determine an end time of the speech, and may provide an indication to the voice activated electronic device of the end time, which in turn may cause the voice activated electronic device to enable the adaptive echo cancellation functionality at the end time.

Abstract: A system for reducing a local oscillator leakage component. The system includes a transmitter channel to transmit data modulated using a transmitter local oscillator frequency. A transmitted signal includes a transmitter local oscillator leakage component. The system also includes a receiver channel to receive the transmitted signal using a receiver local oscillator signal having a frequency offset from the transmitter local oscillator frequency. The received signal includes the transmitter local oscillator leakage component isolated from one or more receiver impairments. The system further includes a feedback loop from the receiver channel to the transmitter channel to identify a power of the isolated transmitter local oscillator leakage component and to generate a local oscillator leakage cancellation signal based on the identified power.

Abstract: An apparatus for detecting fault injection includes functional circuitry and fault detection circuitry. The functional circuitry is configured to receive one or more functional input signals and to process the functional input signals so as to produce one or more functional output signals. The functional circuitry meets a stability condition that specifies that stability of a designated set of one or more of the functional input signals during a first time interval guarantees stability of a designated set of one or more of the functional output signals during a second time interval that is derived from the first time interval. The fault detection circuitry is configured to monitor the designated functional input and output signals, to evaluate the stability condition based on the monitored functional input and output signals, and to detect a fault injection attempt in response to detecting a deviation from the stability condition.

Abstract: An image processing apparatus includes a first controller, a second controller, and a third controller. The first controller controls an output unit that outputs an image. The second controller controls a reading unit that reads an image. The third controller controls, in response to receipt of a power reduction request, the first controller or the second controller to restrict performance of the output unit or the reading unit so as to satisfy the power reduction request.

Abstract: A method implemented during a communication event conducted between a user device and at least one further user device over a communications network, the method comprising: receiving an audio signal that is to be output from a speaker of said user device, said audio signal comprising at least one microphone signal transmitted from the at least one further user device; detecting that said audio signal comprises at least one audio component of a microphone signal transmitted from one or more of one of said at least one further user device that is co-located with said user device; and based on this detection, removing said at least one audio component from said audio signal to produce an output audio signal for output from said speaker.