Abstract: Implementing a circuit design using time-division multiplexing (TDM) can include determining a net signature for each of a plurality of nets of a circuit design. For each net, the net signature specifies location information for a driver and one or more loads of the net. The plurality of nets having a same net signature can be grouped according to distance between drivers of the respective nets. One or more subgroups can be generated based on a TDM ratio for each group. For one or more of the subgroups, a TDM transmitter circuit is connected to a TDM receiver circuit through a selected interconnect, the drivers of the nets of the subgroup are connected to the TDM transmitter circuit, and loads of the nets of the subgroup are connected to the TDM receiver circuit.

Abstract: The application discloses a signal generation circuit (100), configured to generate a transmission signal to trigger a first transducer to generate a first transducer output signal; the signal generation circuit includes: a signal generation unit (106), configured to generate an output signal; and a transmitter (104), coupled to the signal generation unit, wherein the transmitter is configured to convert the output signal into the transmission signal; wherein the transmission signal includes a data signal and a compensation signal, the data signal includes at least one first pulse wave, the compensation signal includes at least one second pulse wave, the first pulse wave and the second pulse wave have opposite phases, and the first pulse wave has an other waveform parameter different from an other waveform parameter of the second pulse wave. The present application further provides a related chip, a flow meter and a method.

Abstract: Echo cancellation in multiple port Full Duplex (FDX) nodes and amplifiers may be provided. First, a plurality of signals may be provided to a cancelation device. Each of the plurality of signals may be configured to provide the cancelation device with a corresponding plurality of configurations. Next, a plurality of effective echo power values respectively corresponding to the plurality of configurations may be received. A one of the plurality of configurations corresponding to a lowest of the plurality of effective echo power values may then be selected. The cancelation device may then be operated at the selected one of the plurality of configurations.

Abstract: A processing system for a transmission apparatus, and an associated transmission apparatus and method, the processing system being configured to produce or generate a transmission signal representative of an original signal comprising data portions; the data being carried by or comprised in waveforms or streams in the transmission signal; the processing system being configured to generate the respective waveforms or streams by loading the data onto selected data symbols or subcarriers in the frequency domain to form the transmission signal.

Abstract: Methods, systems, computer-readable media, and apparatuses for acoustic echo cancellation during playback of encoded audio are presented. In some embodiments, a decoder is arranged to decode an encoded media signal to produce an echo reference signal, and an echo canceler is arranged to perform an acoustic echo cancellation operation, based on the echo reference signal, on an input voice signal to produce an echo-cancelled voice signal. The echo canceler may be configured to reduce, relative to an energy of a voice component of the input voice signal, an energy of a signal component of the input voice signal that is based on audio content from the encoded media signal.

Abstract: A system for detecting divergence in a noise-cancellation system, comprising: a controller configured to: determine a power of a component of the error signal, the component being correlated to the at least one reference sensor signal; determine an average value, over a first time period, of a value representative of a time gradient of the power of the component of the error signal; and determine whether the average value is greater than a threshold.

Abstract: Apparatuses and methods are disclosed for automatic volume control of an audio stream reproduced by a captioning communication service for use by a call assistant in generating a text transcription of a communication session between a hearing-impaired user and a far-end user. The automatic volume control automatically adjusts a volume of the audio stream reproduced by the captioning communication service responsive to a volume control command identifying which of the far-end voice signal and the near-end voice signal is active at a given time. The system further includes an echo modifier configured to add distortion to an echo portion of the far-end voice signal when generating the audio stream.

Abstract: A vibronic sensor for determining and/or monitoring at least one process variable of a medium in a container. The sensor at least comprising: a unit which can oscillate mechanically; a driving/receiving unit; and an electronic unit. The driving/receiving unit is designed to excite, by an electrical excitation signal, mechanical oscillations in the unit which can oscillate mechanically and is designed to receive the mechanical oscillations of the unit which can oscillate mechanically, and to convert the mechanical oscillations into an electrical receiving signal. The electronic unit is designed to generate the excitation signal on the basis of the receiving signal and to determine that at least one process variable from the receiving unit. The electronic unit comprises at least one adaptive filter; and the electronic unit is designed to set the filter characteristic of the adaptive filter in such a way that there is a target phase shift between the excitation signal and the receiving signal.

Abstract: A controller for an acoustic echo canceller is described. The controller includes a noise estimator configured to estimate a level of noise that is comprised in a microphone signal relative to an echo component, estimated by the acoustic echo canceller, comprised in the microphone signal. The controller further includes a control module configured to control the acoustic echo canceller in dependence on that estimate.

Abstract: A system for estimating delay between a far-end signal and an echo of the far-end signal in a microphone signal, the system comprising: a buffer configured to store a time-series of far-end samples representing the far-end signal; a first delay estimator configured to form a first estimate of the delay in respect of a speech frame representing speech in the microphone signal; a second delay estimator configured to form a second estimate of the delay for the speech frame by operating a first set of one or more filters on far-end samples selected from the buffer in dependence on an operating delay; a second set of one or more filters for operation on far-end samples; and a controller configured to, in response to a determination that the first estimate of the delay for the speech frame differs from the operating delay for a previous frame by at least a predefined length of time, cause the one or more filters of the second set to operate on far-end samples selected from the buffer according to the first estimate

Abstract: Disclosed is an in-band full duplex transceiver. The in-band full duplex transceiver may include a transmitter, a hybrid transformer, and an impedance matching unit. The hybrid transformer may include a first end connected to an antenna, may output a transmitting signal to the antenna, and may output a received signal provided through the antenna to the receiver. The impedance matching unit may include balance networks connected to a second end of the hybrid transformer and matching impedance of the antenna.

Abstract: A method and system for generating upstream frames in an Ethernet Passive Optical Network protocol over Coax (EPoC) network is provided. Upstream frame configuration parameters are received by a CNU from a CLT in a link info broadcast a message. The CNU can recover the upstream frame configuration parameters from the link info broadcast message and generate at least in part upstream frame.

Abstract: Noise filling of a spectrum of an audio signal is improved in quality with respect to the noise filled spectrum so that the reproduction of the noise filled audio signal is less annoying, by performing the noise filling in a manner dependent on a tonality of the audio signal.

Abstract: Aspects of the present invention relate to systems, methods and apparatus for identifying a reference audio content in an audio stream.

Abstract: A measuring device serves for the detection of a simultaneous presence of at least a first signal and a second signal in a measurement signal. The measuring device contains a receiving unit for receiving the measurement signal as a digital measurement signal and a processing unit. This processing unit includes a transformation unit for transforming the digital measurement signal into a measurement signal in the frequency domain. The processing unit further contains a subtraction unit for subtracting the first sideband of the measurement signal in the frequency domain from a second sideband of the measurement signal in the frequency domain. In this context, the subtraction unit generates a residual signal.

Abstract: Systems and methods for performing videoconferencing using split endpoints are disclosed herein. These endpoints can include a Main Unit and a Satellite Unit that are associated with each other, and where the tasks of video and audio encoding and decoding are separated between the two. In some embodiments the Main Unit performs video and audio encoding, and the Satellite Unit performs video and audio decoding. In some embodiments the Satellite Unit obtains media data through its associated Main Unit, whereas in other embodiments the Satellite Unit obtains media data directly from the Server.

Abstract: Embodiments include systems and methods for managing an adaptive interference filter performed by a device processor of a mobile communication device. A device processor may determine whether a transmission power from the mobile communication device is greater than or equal to a transmit power threshold. The device processor may determine whether a ratio of interference in a received signal to a level of the received signal including noise (interference ratio) is greater than or equal to an interference ratio threshold in response to determining that the transmission power is greater than or equal to the transmit power threshold. The device processor may perform interference cancellation with the adaptive interference filter in response to determining that the interference ratio is greater than or equal to the interference ratio threshold. In some embodiments the adaptive interference filter may be a non-linear interference cancellation filter.

Abstract: An acoustic echo canceller (AEC) system may be configured to reset the coefficients of a transform equation when an estimated echo diverges from actual acoustic echo. Features are disclosed for determining when to reset the coefficients, and for enabling the reset operation to be performed reliably. Additional features are disclosed for detecting other signal conditions besides AEC divergence, for adjusting the rate at which the coefficients are adapted in response to such conditions, and for prioritizing between potentially incompatible adjustments.

Abstract: Echo removal techniques are described. An echo path of the echo in a received audio signal is modelled using a first model to determine a first model estimate of the echo. The first model estimate is used to determine a first performance value according to a performance metric. The first performance value is compared with a threshold value. It is determined if the echo path can be deemed linear based on the comparison. If so, the first model estimate of the echo is used to remove the echo in the received audio signal. Otherwise, the echo path of the echo is switched to be modelled using a second model to determine a second model estimate of the echo, and the second model estimate of the echo is used to remove the echo in the received audio signal.

Abstract: A universal reconfigurable system and method are provided for reducing nonlinear echo and residual echo, and for echo leakage prevention in various time-varying and complex environments is proposed in this invention. According to one embodiment an echo cancellation system includes (1) an adaptive linear filter implemented in either time-domain or frequency-domain; (2) a nonlinear echo suppression; (3) echo leakage prevention; (4) direct current (DC), low frequency residual echo and noise reduction; (5) time-domain nonlinear processor (NLP) to reduce the residual echo; and (6) frequency-domain NLP to further reduce the residual echo. The echo cancellation system is universally applicable to acoustic echo cancellation (AEC) and/or electrical echo cancellation applications. In terms of AEC, this invention is reconfigurable for one or more microphones and/or one or more reference channels. The numbers of microphones and reference channels are user configurable.

Abstract: An echo canceller in an IP network includes an adaptive filter that models the echo path between a receiving output port of the echo canceller and a sending input port. The adaptive filter filters a receiving input signal to generate an estimate of an echo signal. The estimate of the echo signal is subtracted from a sending input signal to cancel the echo in the sending input signal and to generate a sending output signal. A packet loss detection circuit detects when packet loss occurs in the echo path. Responsive to detection of packet loss in the echo path, the echo canceller applies packet loss concealment to either the sending output signal or the receiving input signal.

Abstract: Features are disclosed for adaptively estimating propagation delay in an audio system. A first device in communication with a speaker may be configured produce sound based on an audio playback signal. A second device in communication with one or more microphones may be configured to detect sound as a microphone signal. The second device may perform acoustic echo cancellation using a first propagation delay parameter and determine a first echo return loss enhancement. The second device may perform acoustic echo cancellation using a second propagation delay parameter and determine a second echo return loss enhancement. A propagation delay between the audio playback signal and the microphone signal may be adaptively estimated based on a comparison of the first and second echo return loss enhancements.

Abstract: An acoustic echo cancellation (AEC) system includes a remote device, for capturing a remote captured sound, a server coupled to the remote device, and a local device coupled to the server. The server transmits the remote captured sound from the remote device to the local device. The local device receives, stores and plays the remote captured sound as a local playback sound. An echo is generated from reflection of the local playback sound. The local device captures the echo and a local sound into a local captured sound, and transmits both the remote captured sound and the local captured sound to the server. The server performs AEC on the local captured sound by using the remote captured sound from the local device and transmits the AEC processed local captured sound to the remote device.

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 packet voice transceiver adapted to reside at a first end of a communication network and to send an ingress communication signal comprising voice packets to, and receive an egress communication signal comprising voice packets from, a second packet voice transceiver residing at a second end of the communication network. The packet voice transceiver includes a far-end echo canceller that reduces echo that is present in the egress communication signal. The far-end communicates with other functional components of the transceiver system and cancels echo or refrains from canceling echo based on the activity of the other functional components.

Abstract: A method and system for improving a perceived duplexity of handsfree telephone applications is disclosed. An echo suppression circuit for a device comprising loudspeaker and microphone is described. A circuit attenuates a subband of a transmit signal, wherein the transmit signal is captured by the microphone and wherein the transmit signal comprises an echo of a far-end signal rendered by the loudspeaker and a near-end signal. The attenuation circuit further determines a subband far-end indicator of a voice activity in the far-end signal; determines a subband near-end indicator of a voice activity of the near-end signal; determines a subband masking weight; determines a subband attenuation for the transmit signal in the subband; and attenuates the subband of the transmit signal using the determined subband attenuation.

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: The present invention provides for echo cancellation circuitry and variable rate encoding circuitry to cooperate with one another to effectively provide comfort noise in an effective and efficient manner. The echo cancellation circuitry will use far-end signals to generate estimated echo signals, which correspond to the actual echo signals appearing in near-end signals. The estimated echo signals are essentially subtracted from the near-end signals in an effort to remove the actual echo signals from the near-end signals. The echo cancellation circuitry will monitor any residual echo signals in the resulting processed near-end signals and provide residual echo control signals that are indicative of whether the residual echo signals should be replaced with comfort noise. The residual echo control signals are used at least in part by the variable rate encoding circuitry to determine the encoding rate to use for encoding different portions of the processed near-end signals.

Abstract: In one embodiment, an acoustic echo control (AEC) module receives an outgoing signal and an incoming signal, which, at various times, contains acoustic echo corresponding to the outgoing signal. The AEC module has a delay estimation block that estimates, in the time domain, the echo delay using an adaptive filtering technique. This delay estimation is used to align samples of the incoming signal having acoustic echo with the corresponding samples of the outgoing signal from which the acoustic echo originated. The AEC module determines whether or not samples of the incoming signal contain acoustic echo based on the aligned outgoing signal, and the determinations are applied to a hangover counter. The AEC module then suppresses acoustic echo in the incoming signal and adds comfort noise to the incoming signal. The amount of echo suppression performed is gradually increased or decreased based on comparisons of the counter to a hangover threshold.

Abstract: The memory storage, transmission and processing demands of a vectored DSL system are reduced by sampling a subset of DSL tones in the DSL tone range used in the vectored system. This data is smoothed (denoised) to further reduce the data's size, sacrificing some fidelity or precision as a result. Finally, lossless entropy coding or the like is performed to encode the FEXT cancellation data for storage and use. The resulting data is less likely to cause transmission bottlenecks in the vectored system, can be stored and used more efficiently for both on-chip and off-chip vectoring implementations, and can be readily updated in various ways.

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: A communication apparatus can perform reliable and stable communication with one or more counterpart apparatuses in a communication network, where the network has plural communication paths in which data communication transmissions are performed according to a predetermined transmission condition. The transmission condition of a communication packet for at least one of the plural communication paths is determined to be different from the predetermined transmission condition if bandwidth measurement is not performed on that communication path while bandwidth measurement is performed on one or more of the other communication paths. Accuracy of measuring available bandwidth in the communication network can be improved because communication paths that perform bandwidth measurements are less impacted by communication paths that do not perform bandwidth measurement.

Abstract: Various embodiments of a wireless multimode receiver having an off-chip duplex filter associated with a multimode band, and a blocker cancellation circuit disposed on a semiconductor chip are described in the present disclosure.

Abstract: A transceiver system is disclosed. The transceiver system comprises a first transceiver physical layer circuit (PHY) having a first plurality of channels and a second transceiver PHY disposed adjacent the first transceiver PHY and having a second plurality of channels. Filter circuitry is coupled between at least one of the plurality of first channels and at least one of the plurality of second channels.

Abstract: A wireless repeater includes a receiver circuit implemented as an intermediate frequency (IF) subsampling receiver and a transmitter circuit implemented as a direct conversion (zero-IF) transmitter. The repeater further includes a reference receiver implemented as an IF subsampling receiver to sample a portion of the transmit signal prior to over-the-air transmission for use as the reference signal for channel estimation. Highly accurate channel estimation is obtained by using the reference signal from the reference receiver as the reference signal accounts for distortions in the transmitter circuit of the repeater. The repeater may include an echo canceller to cancel an estimated feedback amount from an input signal based on the channel estimation.

Abstract: A system and method for provide a stable gain from an adaptive gain control device in a signal path. An echo canceller is also located in the signal path, and is used to provide performance information regarding losses in the signal. This performance information is fed to the automatic gain control device via a connection. The automatic gain control device thereafter uses the performance information to determine a maximum gain that might be provided based upon losses cause by echo conditions. The gain however is limited in order to provide for a stable system. The performance information includes a loss rate that includes a combination of the echo return loss and the echo return loss enhancement.

Abstract: Provided are systems and methods for improving audio signal quality during teleconferencing. Multiple local devices equipped with microphones may be connected to a teleconference hub device that also maintains communication with one or more remote devices. The local devices may be mobile, such as mobile phones, and each may be positioned closer to at least some local teleconference participants compared to a microphone of a traditional central teleconferencing speakerphone. Such mobility allows users to move during the teleconference without disturbing the quality. The local devices may be connected to the hub device using various local protocols. Individual audio signals may first be processed by the local devices and then sent to the hub device for further collective processing of multiple data streams and production of an external data stream, which is sent to remote devices connected to the hub device over a remote protocol.

Abstract: A packet voice transceiver adapted to reside at a first end of a communication network and to send an ingress communication signal comprising voice packets to, and receive an egress communication signal comprising voice packets from, a second packet voice transceiver residing at a second end of the communication network. The packet voice transceiver includes a far-end echo canceller that reduces echo that is present in the egress communication signal. The far-end communicates with other functional components of the transceiver system and cancels echo or refrains from canceling echo based on the activity of the other functional components.

Abstract: An echo cancellation circuit in a full duplex two-way communication system comprising: an input/output terminal; a subtractor having a positive and a negative input terminals, in which a first transmission signal is inputted to the negative input terminal as a pseudo echo signal, the first transmission signal is inputted through an output buffer to the positive input terminal as an echo signal, the pseudo echo signal inputted to the negative input terminal is subtracted from the echo signal inputted to the positive input terminal; and a result of the subtraction is outputted; and an echo cancellation error reducing unit having a D/A converter at an input side or an output side of the subtractor.

Abstract: A signal format that can be accessed, as a file, by the computer on which it is stored is provided with a plurality of concatenated content packages. The content package comprises System, Picture, Audio and Auxiliary Items. The System, Picture, Audio and Auxiliary Items have a common format including a 16 byte SMPTE Universal Label, a Word Count, and Data. The Item Label has a predetermined length which is set to fixed value except for the byte identifying the Item to which the label belongs. The data of the System Item includes a set of Metadata blocks, which has an initial Metadata Count byte defining the number of blocks in the set. A metadata block may include a metadata Link which links the metadata of that block to the essence of the element of the Auxiliary Item with which element it is associated.

Abstract: A network management/apparatus connected to a plurality of packet relay apparatuses constituting a group through a network includes an energy saving level calculation unit for calculating an energy saving level of each of the packet relay apparatuses on the basis of a CPU usage ratio of the packet relay apparatus managed or a difference between the present energy saving level and the energy saving level of minimum guarantee of the packet relay apparatus, and an energy saving function setting unit for setting an energy saving function of the packet relay apparatuses on the basis of the energy saving level calculated by the energy saving level calculation unit.

Abstract: A router of a first network and a router of a second network are connected to a third network. A connection request including information relating to the direction of communication performed by a first service provider that exists on the first network is transmitted from the first network to the second network via the third network. On the basis of the information relating to the direction of communication included in the connection request transmitted from the first network to the second network, a second service provider that exists on the second network is selected and a response that includes identification information of the selected second service provider is transmitted from the second network to the first network via the third network.

Abstract: A method is provided which is suitable to cope with non-linear echo paths during acoustic echo cancellation in speakerphones. Non-linear paths occur particularly in hands-free operation of, e.g., a mobile phone, due to driving the amplifier and loudspeaker in the non-linear range. The idea is to combine the commonly known one microphone approach of linear acoustic echo cancellation using an adaptive filter and a post-processor together with a multiple microphone approach using beam forming which separately removes the non-linear part of the echo.

Abstract: A system, method, and computer program product are provided for processing different content each stored in one of a plurality of queues. In use, a plurality of different content is identified for processing. Additionally, each of the different content is stored in one of a plurality of queues based on a classification thereof. Furthermore, the plurality of different content stored in the plurality of queues is processed.

Abstract: Techniques to characterize and reduce acoustic echo are described. For example, a mobile computing device may comprise an audio characterization module to send a first signal to a first component and receive a second signal from a second component. The first signal may be a deterministic audio training sequence and the second signal may be a recovered audio training sequence. The audio characterization module compares the first signal and the second signal to determine an audio echo path. Other embodiments are described and claimed.

Abstract: A method for performing a call between a near-end user and a far-end user, which includes the following operations performed during the call by the near-end user's communications device. Automatic gain control (AGC) is performed to update a gain applied to an uplink speech signal. A frame is detected in a downlink signal that contains speech; in response, the updating of the gain is frozen. Other embodiments are also described and claimed.

Abstract: An echo suppressing apparatus operable with a sound output device for outputting sound based on a reference signal and a sound processing device for processing an observed signal based on a received sound, the echo suppressing apparatus includes: an echo removal section for removing an echo component of the observed signal which is presumed on the basis of both the observed signal and the reference signal; a ratio calculation section for calculating a power ratio of the observed signal to the reference signal; a correction amount calculation section for calculating, based on the power ratio, a correction amount for the observed signal after removal; a correction section for correcting, based on the calculated correction amount, the observed signal after removal, so as to suppress a residual echo remaining in the observed signal after removal; and an output section for outputting the corrected observed signal to the sound processing device.

Abstract: A divergence detection device for an adaptive system comprising means for calculating the energy of an input signal of the system, means for calculating the energy of an output signal of the system and means for analysing the energy of the input signal and the energy of the output signal so as to detect a rise in the energy of the output signal, relative to the energy of the input signal, in case of divergence of the system.

Abstract: A system, method, and program product are provided for filtering sound from a selected application on a computer without interrupting voice communications on the computer. The method comprises: monitoring a selected program for an outgoing digital audio signal from a selected application; detecting said digital audio signal; and filtering an analog microphone input with the digital audio signal.

Abstract: Sound signals captured by a microphone are adjusted to provide improved sound quality. More particularly, an Acoustic Echo Reduction system which performs a first stage of echo reduction (e.g., acoustic echo cancellation) on a received signal is configured to perform a second stage of echo reduction (e.g., acoustic echo suppression) by segmenting the received signal into a plurality of frequency bins respectively comprised within a number of frames (e.g., 0.3 s to 0.5 s sound signal segments) for a given block. Data comprised within respective frequency bins is modeled according to a probability density function (e.g., Gaussian distribution). The probability of whether respective frequency bins comprise predominantly near-end signal or predominantly residual echo is calculated.