Abstract: The present technology relates to an information processing apparatus, an information processing method, and a program which can curb occurrence of howling at the time of outputting vibration in response to an input sound. The information processing apparatus of one aspect of the present technology is an apparatus that generates, at the time of outputting vibration in response to an input sound from the outside, a vibration signal representing the vibration having a frequency different from a frequency of the input sound. The present technology can be applied to, for example, smartphones, smart watches, wearable apparatuses, cushions, and music experience apparatuses that vibrate in response to input sounds.

Abstract: Echo cancellation for a two-way audio communication includes receiving, at an AEC system from microphone(s), an audio signal based on, at least in part, near-end signals and reproduced far-end signals. Loudspeaker(s) reproduced the far-end signals. The AEC system is operated, at least in part, with filter(s) so as to update estimates of coefficients of an acoustic channel from the loudspeaker(s) to the microphone(s). Control parameter(s) affecting an operation of the AEC system that is/are configurable and is/are set to value(s), from a range of values, is/are determined, based on estimating an accuracy of the estimates of the coefficients of the acoustic channel and a characteristic of the near-end signals. The AEC system controls the filter(s) with different values of the control parameter(s) at different times.

Abstract: An acoustic noise suppressing apparatus includes a sound pickup circuit, a first and second suppression circuits, and an output signal selection circuit. The sound pickup circuit picks up sound. The first suppression circuit processes the sound, in which the first suppression circuit is configured to calculate a first suppression sound signal in which acoustic noise is suppressed from the sound by using a first algorithm suitable for multiple sound sources. The second suppression circuit processes the audio signal in parallel with the first suppression circuit, in which the second suppression circuit is configured to calculate a second suppression sound signal in which acoustic noise is suppressed from the sound signal by using a second algorithm suitable for a single sound source. The output signal selection circuit outputs only one of the first suppression audio signal and the second suppression audio signal.

Abstract: An electronic device includes a microphone array to capture audio input data, a speaker array to render audio output data for playback; one or more sensors to detect an orientation of the microphone array, acoustic echo cancellation logic, and an interface. The acoustic echo cancellation logic applies acoustic echo cancellation to the audio input data to form echo-cancelled audio input data based on the orientation of the microphone array. The interface transmits the echo-cancelled audio input data over a communications channel as part of an audiovisual communication system.

Abstract: A physical layer transceiver, for connecting a host device to a wireline channel medium in which a signal component occurring at a particular time may cause interference at other times, includes a host interface for coupling to a host device, a line interface for coupling to the channel medium, and filter circuitry operatively coupled to the line interface to filter the interference caused by the signal component at the particular time and at the one or more other times. The filter circuitry includes at least one filter segment configured to operate at a first rate derived from a channel operating frequency to filter the signal component at the particular time, and at least one respective filter segment configured to operate at a respective additional rate different from the first rate. Respective delay elements allow each respective filter segment to filter the signal component at a one of the other times.

Abstract: A system and method for providing a headset system including a microphone including a microphone transducer configured to receive vocal sounds from a user of the headset system and to generate a voice audio signal representing the vocal sounds; a first earcup including a first housing, wherein the first housing further includes a voice canceling circuit, an inner speaker, and a first outer speaker; the voice canceling circuit configured to receive the voice audio signal and configured to generate a first voice canceling signal to cancel the voice of the user; and the first outer speaker including a first speaker transducer configured to generate and output sounds representing the first voice cancelling signal to cancel the vocal sounds of the user.

Abstract: An acoustic noise suppressing apparatus outputs a first suppression audio signal in which the acoustic noise is suppressed by subtracting a first pseudo noise signal from the picked up audio signal, the first pseudo noise signal being generated based on a first delay signal and a first filter updated by a first algorithm which is valid when a plurality of talkers are talking, and outputs a second suppression audio signal in which the acoustic noise is suppressed by subtracting a second pseudo noise signal from the picked up audio signal, the second pseudo noise signal being generated based on a second delay signal and a second filter updated by a second algorithm which is valid when one talker is talking. The apparatus outputs a suppressed one of the first suppressed audio signal or the second suppressed audio signal.

Abstract: A multi-line digital transceiver configured to use digital signal vectoring in a manner that causes effects of crosstalk between distinct groups of subscriber lines to be effectively mitigated, without directly attempting to mitigate effects of crosstalk within any one of those distinct groups. In an example embodiment, effects of crosstalk within each of the distinct groups can be mitigated indirectly using an appropriate T/FDMA schedule, according to which, during a given symbol period, a given resource block of any of the distinct groups can carry data corresponding to a single respective subscriber. A precoder (postcoder) matrix for the digital signal vectoring can be generated using block-diagonalization techniques appropriately constrained, e.g., using the groups' definitions, aggregate-transmit-power restrictions, etc. In various embodiments, the disclosed digital signal vectoring can be used on the downlink or on the uplink, or both.

Abstract: A microphone picks-up voice of a driver. A first echo suppression unit outputs a voice signal after first echo suppression based on a voice signal of the driver and a voice signal after echo suppression in the past (first reference signal) stored in a buffer memory. A second echo suppression unit outputs a voice signal after second echo suppression based on a voice signal of the driver and a voice signal after the echo suppression in the past (second reference signal) stored in a buffer memory. An output signal selector selects one of the voice signals after the first echo suppression or the voice signal after the second echo suppression according to a detection result of the presence or absence of a system variation by a system variation detector, and causes a speaker to output the selected voice signal.

Abstract: A system for an integrated voice and data communications environment is disclosed. The system may receive a voice call and assign the voice call to a first communications environment. The system may store call related data from the first communications environment. The system may receive a request for communications transfer and transfer the voice call from the first communications environment to a second communications environment in response to the request. The system may pass the call related data from the first communications environment to the second communications environment.

Abstract: Methods and devices are provided for a wireless microphone network whereby robustness for reception of audio information transmitted by one or more wireless microphones is enhanced. The system incorporates a dual stage network approach for collecting, transmitting and receiving audio information. In the first stage, audio information collected by one or more microphone modules is transmitted to a series of receiver base stations. In the second stage, information received at each base station is subsequently transmitted to a receiver hub that selects or augments the information to produce a high quality representation of audio information, which provides improved robustness and reliability with respect to the movement or physical placement of each microphone module.

Abstract: A method detects whether an ear mold of a listening device is correctly mounted in the ear of a user. An indication of whether or not a mold of a listening device is correctly mounted in an ear canal of a user is provided. The method comprises a) providing a long term estimate of the feedback path; b) providing an estimate of the current feedback path; c) comparing the long term feedback path estimate with the current feedback path estimate, and providing a measure of their difference, termed the feedback difference measure FBDM; and optionally d) providing an alarm indication, if the feedback difference measure exceeds a predefined threshold.

Abstract: A system and method to perform an estimation of a location of an active speaker in real time includes designating a microphone of an array of microphones as a reference microphone. The method includes storing a relative transfer function (RTF) for each microphone of the array of microphones other than the reference microphone associated with each potential location among potential locations as a set of stored RTFs, and obtaining a voice sample of the active speaker and obtaining a speaker RTF for each microphone of the array of microphones other than the reference microphone. The method also includes performing an RTF projection of the speaker RTF for each microphone on the set of stored RTFs, and determining one of the potential locations as the location of the active speaker based on the performing the RTF projection.

Abstract: Embodiments include systems and methods for implementing a multi-rate FIR by using rate-dependent bit stuffing on the cursor, while using rate-independent (e.g., full-rate) spacing on the pre- and post-cursor. For example, in the FIR data path, the cursor bit output is generated using bit stuffing, depending on a selected rate mode (e.g., full-rate, half-rate, quarter-rate, eighth-rate, etc.), but the spacing of the pre-cursor, cursor, and post-cursor are maintained at 1 UI apart (i.e., the full-rate spacing) for all rate modes. Such an approach can appreciably reduce complexity of the logic and can appreciably relieve the critical timing path.

Abstract: A system and method for acoustic management that includes improving the sound quality of two or more audio processing modules in an acoustic environment may receive first control parameters from a first audio processing module. Receiving second control parameters from a second audio processing module. An audio processing interaction may be derived between with the first audio processing module and the second audio processing module determined from the first control parameters and the second control parameters. The first control parameters and the second control parameters may be modified responsive to the derived audio processing interaction.

Abstract: Apparatus and methods concerning simulation of call quality are disclosed. In an example embodiment, computing server is communicatively coupled to a VoIP server. The computing server includes a communication circuit configured to receive a first set of VoIP data including audio of a VoIP call routed by the VoIP server. The computing server also includes a processing circuit configured to characterize a post-transmission quality state of the first set of VoIP data. The processing circuit is also configured to generate a second set of VoIP data including audio that is different from the audio of the VoIP call and data including characteristics indicative of the post-transmission quality state of a first set of VoIP data. The processing circuitry may configured to use the second set of data to provide security, protect the confidentiality and privacy, and/or monitor changes of behavior/quality for different audio CODECs, encryption, bit-rate, etc.

Abstract: An off-head detection system for an in-ear headset comprises an input device that receives an audio signal, a feed-forward microphone signal, and a driver output signal; an expected-output computation circuit that predicts a value of the driver output signal based on a combination of the audio signal and the feed-forward microphone signal from the signal monitoring circuit, and off-head data from the off-head model; and a comparison circuit that compares the observed output signal provided to the driver and the computed expected output to determine an off-head state of the in-ear headset.

Abstract: A system and method for acoustic management that includes improving the sound quality of two or more audio processing modules in an acoustic environment may receive first control parameters from a first audio processing module. Receiving second control parameters from a second audio processing module. An audio processing interaction may be derived between with the first audio processing module and the second audio processing module determined from the first control parameters and the second control parameters. The first control parameters and the second control parameters may be modified responsive to the derived audio processing interaction.

Abstract: An energy harvester having a converter suitable for converting a variation of the energy to be harvested into a corresponding excess of electrical charges. There is a circuit for collecting the excess of electrical charges, the circuit is equipped with a controllable switch and a control device for the switch designed to control the switching of this switch to its closed position. The control device is suitable for exerting a force which stresses the electrical contacts towards one another. The force varies continually as a function of the quantity of electrical charges present on the first terminal and bringing the electrical contacts to bear on one another only when the excess of electrical charges exceeds a predetermined threshold, and/or for ionizing the electrically insulating medium to produce an electrical arc between the two electrical contacts only when the excess of electrical charges exceeds the predetermined threshold.

Abstract: An echo path monitoring system for controlling an adaptive filter configured to estimate an echo of a far-end signal comprised in a microphone signal, the system comprising a comparison generator configured to compare the microphone signal with the estimated echo to obtain a first comparison and compare an error signal, which represents a difference between the microphone signal and the estimated echo, with the estimated echo to obtain a second comparison, and a controller configured to combine the first and second comparisons to form a parameter indicative of a state of the microphone signal and, in dependence on said parameter, control an operating mode of the adaptive filter.

Abstract: Method(s) and system(s) for generation of randomized messages for cryptographic hash functions are described herein. The method includes obtaining a random value based on a randomization criterion to randomize a message. Further, a last data block of the message is populated with a randomization parameter to obtain a randomized message. The randomization parameter populated in the last block is computed using the random value.

Abstract: A system including a processor that receives a first digital signal corresponding to a first signal received by the system, and a second digital signal corresponding to a second signal transmitted by the system; generates a first control signal based on the first and second digital signals; and generates a first output signal based on the first control signal and a second analog signal corresponding to the second signal. A cancellation device generates a second control signal based on a first analog signal corresponding to the first signal, and a third control signal based on the first control signal and the first output signal; selects the second or third control signal when the system operates in a first or a second mode; and outputs a second output signal to cancel interference in the first signal according to the second analog signal and one of the second and third control signals.

Abstract: Features are disclosed for performing noise compensation so a level of noise may be audible in an output signal. For example, a first noise level of a first signal can be estimated. The first signal can be processed (e.g., by residual echo suppression) to determine a second signal, and a second noise level of the second signal can be estimated. Residual echo suppression can sometimes cause background noise to be eliminated, causing silence. If the second noise level is less than a product of the first noise level and a noise threshold, then a noise signal based on the first noise level can be determined, wherein the noise level of the noise signal is configured to be above a hearing threshold. The noise signal can be combined with the second signal to generate an output signal.

Abstract: Provided is a system for adaptively enhancing an end-user's perceived quality, or quality of experience (QoE), of speech and other audio under ambient noise conditions. The system comprises the steps of determining the ambient noise characteristics on a continuous basis to capture the time varying nature of ambient noises, and adaptively determining the most optimal signal shaping to be applied to the audio/speech signal to produce the most appropriate enhancement to compensate for the ambient noise impairment. The system also comprises a signal shaping technique by using an infinite impulse response (IIR) filter that performs the signal modification with a low delay; a multi-level automatic gain control (AGC); and a controlled amplitude clipping module that assures samples are below a certain limit; and outputs the modified signal for playback through a loudspeaker or the like.

Abstract: In one implementation, a conference hosted on a conference bridge that includes an echo control device that classifies the echo return loss of a plurality of endpoints. The endpoints may include endpoints on internet protocol (IP) networks and public switched telephone networks (PSTN). The echo control of the endpoints is controlled independently. An echo classification module calculates the echo return loss value of the associated echo tail of the endpoint and chooses an appropriate echo control mode. If the echo return loss is above a high threshold, a pass through mode is activated. If the echo return loss is below a low threshold, a full echo cancellation mode is activated. If the echo return loss is between the low threshold and the high threshold, an echo suppression mode is activated.

Abstract: Clock synchronization for an acoustic echo canceller (AEC) with a speaker and a microphone connected over a digital link may be provided. A clock difference may be estimated by analyzing the speaker signal and the microphone signal in the digital domain. The clock synchronization may be combined in both hardware and software. This synchronization may be performed in two stages, first with coarse synchronization in hardware, then fine synchronization in software with, for example, a re-sampler.

Abstract: The present invention relates to methods and apparatuses for extending the benefits of vectoring technology to xDSL systems including disparate types of xDSL lines. In embodiments, the invention includes a scheme to match the symbol frequencies of ADSL2 and VDSL2 lines, to then enable vectoring of ADSL2 FEXT into VDSL2 lines.

Abstract: A compensator generating a compensation signal to compensate for nonlinear echo in an output of a current source. The nonlinear echo is a result of transitioning the current source between an ON state and an OFF state. The compensator includes driving, weighting, function, and compensating circuits. The driving circuit receives a first signal that is based on the output of the current source. The weighting circuit is configured to generate a second signal based on weighted versions of the first signal. The function circuit, based on the second signal, (i) updates each of multiple functions, and (ii) selects a first function. The driving circuit generates a driving signal based on the first function selected by the function circuit. The compensating circuit generates the compensation signal based on the driving signal to compensate for the nonlinear echo provided by the output of the current source.

Abstract: A step size controller may be used to control the rate of adaptation in an acoustic echo canceller. Step size control based on the values of adaptive coefficients (rather than, e.g., a fixed initial adaptation period) provides improved reliability and resistance to disruption. Accordingly, features are disclosed for controlling step size based on the values of adaptive coefficients.

Abstract: An apparatus having a first circuit and a second circuit is disclosed. The first circuit is configured to generate a first intermediate signal by expanding a first input signal subjected to a quantization. The second circuit is configured to generate a second intermediate signal based on a second input signal. The second intermediate signal approximates an echo in the first input signal caused by the second input signal. The second circuit is also configured to generate a third intermediate signal by companding the second intermediate signal to compensate the quantization of the first input signal.

Abstract: In an embodiment, the access node includes first and second transceiver units for initializing first and second communication channels, respectively, over first and second communication lines, respectively. The initialization of the first and second communication channels respectively including first and second handshake phases wherein peer transceiver units exchange capabilities and select (i) a common mode of operation, (ii) first and second crosstalk channel acquisition phases from the first and second communication lines, and (iii) first and second generic initialization phases. The access node further includes a line initialization controller configured to stretch the execution of the first handshake phase with respect to the execution of the second handshake phase so as to shorten a time difference between respective entries into the first and second crosstalk channel acquisition phases.

Abstract: A sine wave generating device includes at least one adder configured to add two input signals thereof, at least one delay unit configured to delay an input signal thereof by one sample time and at least one multiplier configured to receive the delayed signal from the at least one delay unit, multiply the delayed signal by a coefficient and output the multiplied signal to provide to the at least one adder. The coefficient is arbitrarily set from outside the sine wave generating device.

Abstract: A method and a device for data processing in a digital subscriber line environment, wherein a flat noise margin is determined based on an actual noise determined and based on a maximum noise expected. A device is initialized using the flat noise margin. Also, a communication system is described that includes such a device.

Abstract: Residual FEXT resulting from intended and/or inherent partial cancellation of crosstalk in vectored DSL systems impairs upstream power back-off (UPBO) as traditionally implemented. By considering and taking into account the effects of residual crosstalk on vectored DSL system performance and operation, improved data rates and/or other vectored DSL system performance are realized through the use of UPBO parameters generated for a given residual FEXT environment.

Abstract: A method of echo cancellation in an acoustic system, comprising a first user device and at least one further user device, the method comprising, at the first user device: detecting an audio signal received from the at least one further user device over a communications network; supplying the audio signal when received to an audio signal processing module of the first user device, wherein the audio signal processing module processes the audio signal and outputs a processed audio signal to audio output means of the first user device; and controlling an echo cancellation process for cancelling echo from an audio signal received via audio input means of the first user device based on a loopback signal provided by the processed audio signal and selectively based on the received audio signal when detected.

Abstract: An encoding device, a decoding device, and related methods are provided that eliminate the loss of synchronization of the adaptive filters of a terminal at the encoding end and a terminal at the decoding end caused by transmission errors. Deterioration of the sound quality is suppressed when a multiple channel signal is encoded with high efficiency using an adaptive filter. In the terminal at the encoding end, a buffer stores updated filter coefficients. When packet loss detection information indicating whether there is any packet loss in the terminal at the decoding end indicates that there is packet loss, a switch outputs the past filter coefficients of the previous (NX+1) frames from the buffer to an adaptive filter. The adaptive filter uses the past filter coefficients of the previous (NX+1) frames to conduct filtering.

Abstract: Embodiments of the present invention disclose a method and a related device and system for crosstalk cancellation processing on a port line of an access device. The method for crosstalk cancellation processing on a port line of an access device includes: obtaining line configuration information of another access device connected to—a first access device, and line group identities of different port lines belonging to a same cable bundle are the same; respectively obtaining, by the first access device, according to the line configuration information of the another access device, a coefficient of an effect from a port line of the another access device having a same line group identifier as that of a local port line on the local port line; and performing crosstalk cancellation processing on the local port line on the basis of the obtained effect coefficient.

Abstract: In response to a first envelope within a kth frequency band of a first channel, a speech level within the kth frequency band of the first channel is estimated. In response to a second envelope within the kth frequency band of a second channel, a noise level within the kth frequency band of the second channel is estimated. A noise suppression gain for a time frame n is computed in response to the estimated speech level for a preceding time frame, the estimated noise level for the preceding time frame, the estimated speech level for the time frame n, and the estimated noise level for the time frame n. An output channel is generated in response to multiplying the noise suppression gain for the time frame n and the first channel.

Abstract: An acoustic echo suppression unit according to an embodiment of the present invention includes and input interface for extracting a downmix signal from an input signal, the input signal including the downmix signal and parametric side information, wherein the downmix and the parametric side information together represent a multichannel signal, a calculator for calculating filter coefficients for an adaptive filter, wherein the calculator is adapted to determine the filter coefficients based on the downmix signal and a microphone signal or a signal derived from the microphone signal, and an adaptive filter adapted to filter the microphone signal or the signal derived from the microphone signal based on the filter coefficients to suppress an echo caused by the multichannel signal in the microphone signal.

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

Abstract: An echo cancelling device includes an adaptive filter which synthesizes a spurious echo signal from a receiving signal, a first subtractor which generates an echo cancellation signal based on an input signal including a voice signal of a speaker and the spurious echo signal, and an error signal generating circuit which generates an error signal based on the input signal and the echo cancellation signal.

Abstract: A vector DSL system includes a plurality of modems, which may be multi-port devices. Unprocessed user data is extracted from the modems and passed through a private vectoring data routing apparatus to one or more vectoring modules, such as vectoring cards. Each vectoring module includes one or more vector processors that include processing units configured to process the unprocessed user data on the basis of all modems' data for a given DSL tone grouping. Processing of the unprocessed user data removes the effects of FEXT from upstream and downstream user data and returns the processed user data to the modems using the vectoring data routing apparatus, which can be a specialized data transmission network utilizing one or more vector routers.

Abstract: Residual FEXT resulting from intended and/or inherent partial cancellation of crosstalk in vectored DSL systems impairs upstream power back-off (UPBO) as traditionally implemented. By considering and taking into account the effects of residual crosstalk on vectored DSL system performance and operation, improved data rates and/or other vectored DSL system performance are realized through the use of UPBO parameters generated for a given residual FEXT environment.

Abstract: A telecommunication system including a fall duplex speakerphone, comprising a first microphone to generate a coupled signal including uplink information and non-linear distortion sensed by the first microphone in a speaker phone mode, a second microphone to generate a reference signal including downlink information and the non-linear distortion sensed by the second microphone in the speaker phone mode, and an acoustic echo canceller (AEC) to receive the coupled signal from the first microphone, to receive the reference signal from the second microphone, and to cancel out the non-linear distortion included in the coupled signal based on the non-linear distortion included in the reference signal.

Abstract: An echo canceler circuit (10) and method attenuates at least post-echo canceler uplink data (90) to produce attenuated uplink data (100) in response to uplink echo return loss based attenuation data (40). The echo canceler circuit (10) includes an echo return loss based attenuation data generator (20) and at least an uplink data attenuator (30). The echo return loss based attenuation data generator (20) produces the uplink echo return loss based attenuation data (40) in response to echo return loss data (70). The echo return loss data (70) is based on at least one of: attenuated downlink data (50), pre-echo canceler uplink data (60), and/or amplifier gain data (80). The uplink data attenuator (30) attenuates the post-echo canceler uplink data (90) to produce attenuated uplink data (100) based on the uplink echo return loss based attenuation data (40).

Abstract: A method and an audio processing system determine a system parameter, e.g. step size, in an adaptive algorithm, e.g. an adaptive feedback cancellation algorithm so as to provide an alternative scheme for feedback estimation in a multi-microphone audio processing system. A feedback part of the system's open loop transfer function is estimated and separated in a transient part and a steady state part, which can be used to control the adaptation rate of the adaptive feedback cancellation algorithm by adjusting the system parameter, e.g. step size parameter, of the algorithm when desired system properties, such as a steady state value or a convergence rate of the feedback, are given/desired. The method can be used for different adaptation algorithms such as LMS, NLMS, RLS, etc. in hearing aids, headsets, handsfree telephone systems, teleconferencing systems, public address systems, etc.

Abstract: At least one example embodiment discloses a method of reducing crosstalk interference in a system having a plurality of communication paths. The method includes obtaining a plurality of signal values corresponding to a set of active communication paths, the active communication paths being a subset of the plurality of communication paths, obtaining an inverse channel matrix, the inverse channel matrix representing an inverse crosstalk channel matrix of the system when the plurality of communication paths in the system are active, determining a plurality of sub-matrices from the inverse channel matrix based on the set of active communication paths and compensating the signal values for crosstalk by individually applying at least one of the sub-matrices to the plurality of signal values.

Abstract: An apparatus comprising a receiver coupled to a digital subscriber line (DSL) between an exchange site and a customer premise equipment (CPE) and configured to send a feedback error message to train a precoder coupled to the exchange site, wherein the feedback error message comprises a plurality of error components and an indication of a quantity of bits per error component, a quantization accuracy per error component, or both. Included is a method comprising sending an error feedback message to a DSL crosstalk precoder to train the crosstalk precoder, wherein the error feedback message comprises an error vector and a quantization scaling factor of the error vector.

Abstract: A nonlinear echo compensator comprises a mapping circuit that includes a weighting circuit that generates a weighted signal based on a current symbol and a prior symbol and a function generating circuit that selects one of N functions based on the weighted signal, where N is an integer greater than one. The mapping circuit generates a driving signal based on the selected one of the N functions and the weighted signal. A canceling circuit generates a nonlinear echo compensation signal based on the driving signal.

Abstract: An echo cancellation device relies on the known characteristics of the sync frame to monitor, update in an off-line fashion and determine the accuracy of an echo canceller in, for example, a modem, such as an ADSL modem. Specifically, time domain samples are read from the transmit (Tx) and receive (Rx) paths of the modem. These samples are stored in memory. When the sync frame has received a predetermined number of the same Tx samples and Rx samples, the samples are stored. Running averages, over the sync frames, of the TX and RX samples are maintained. These averages are subtracted from a sync frame of samples, to allow LMS updating of the echo canceller taps, free of extraneous signals. Updating, i.e., tracking of changes in the echo channel, is done for the echo canceller in an off-line fashion. The coefficients for the in-line version are updated, while the off-line version is updated over several sync frames. Periodically, the performance of the off-line version is compared with the in-line version.