Abstract: In one embodiment, a two-way telecommunication device may perform acoustic echo cancellation on incoming signals. An audio decoding module may produce an audio render signal. An audio capture interface may receive an audio capture signal. A short length adaptive filter may determine a time delay between the audio render signal and the audio capture signal by adaptively predicting a sub-band of the audio capture signal using a corresponding sub-band of the audio render signal.

Abstract: Processing requirements for acoustic echo cancellation in voice communications are significant and are even more so as the bandwidth of the communication increases. Whilst voice communication occupies a relatively narrow band of frequencies the processing requirements and so forth for wideband communication render acoustic echo cancellation difficult to achieve in a cost effective manner. The invention provides for acoustic echo cancellation within wideband communications by dividing the communications into sub-bands and applying acoustic echo cancellation to some sub-bands whilst processing other sub-bands according to the status of the communications. Additional sub-bands are transmitted at either full-duplex or half-duplex.

Abstract: An echo canceller used for hands-free communication systems in which hands-free communication is performed by using a speaker and a microphone is disclosed. The echo canceller includes a step size control unit calculating a step size value in an adaptive filter and an adaptive filter unit estimating an echo component of a feedback path from an input signal to the feedback path by adaptively identifying an impulse response of the feedback path formed by an acoustical coupling and the like of the speaker and the microphone, and subtracting the echo component from an output signal from the feedback path, in which the step size control unit calculates a step size value by using an echo reduction amount defined based on the ratio between the output signal from the feedback path and a residual signal and outputs the value to the adaptive filter unit.

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: A method and corresponding apparatus for coded-domain acoustic echo control is presented. An echo control problem is considered as that of perceptually matching an echo signal to a reference signal. A perceptual similarity function that is based on the coded spectral parameters produced by the speech codec is defined. Since codecs introduce a significant degree of non-linearity into the echo signal, the similarity function is designed to be robust against such effects. The similarity function is incorporated into a coded-domain echo control system that also includes spectrally-matched noise injection for replacing echo frames with comfort noise. Using actual echoes recorded over a commercial mobile network, it is shown herein that the similarity function is robust against both codec non-linearities and additive noise. Experimental results further show that the echo-control is effective at suppressing echoes compared to a Normalized Least Mean Squared (NLMS)-based echo cancellation system.

Abstract: An echo detector includes means (34) for forming a set of distance measures between pitch estimates of a first signal and pitch estimates of a second signal at predetermined delays with respect to the first signal. A selector (36) selects a distance measure from the set corresponding to the highest similarity between the first and second signals. A classifier (32) classifies the second signal as including an echo if the selected distance measure has a predetermined relation to a similarity threshold (TH).

Abstract: A feedback calibration system and a method for controlling an electronic signal are disclosed. The feedback calibration system includes an input controller adapted to modify an input signal in response to a control signal and generate a modified input signal, a signal processing block including a signal analyzer, wherein the signal processing block is adapted to process the modified input signal to generate an output signal and the signal analyzer is adapted to detect an undesirable condition of the output signal and transmit a detection signal corresponding to the undesirable condition, a transfer function estimator adapted to model and transmit a transfer function estimate of the signal processing block in real-time in response to the detection signal, and a programmable device adapted to transmit the control signal to the input controller for modifying the input signal, wherein the control signal is based upon the transfer function estimate.

Abstract: An apparatus to enable half-duplexing capabilities in a two-way communication device is disclosed. The apparatus estimates the signal power and background noise of a first input signal and a second input signal during approximately the same period. The apparatus further provides at least one control signal based on the result of one or more determinations. These determinations may include whether the estimated signal power of at least one of the first and second input signals exceeds a threshold value; whether the estimated signal power of the first input signal exceeds the sum of a first threshold value and the estimated background noise of the first input signal; and whether the estimated signal power of the second input signal exceeds the sum of a second threshold value and the estimated background noise of the second input signal. Other embodiments for use with two-way communication devices and related methods are also disclosed.

Abstract: A method and apparatus for reducing crosstalk between digital subscriber lines includes: firstly, determining a parameter of VDSL (Very-high-bit-rate Digital Subscriber Line); secondly, determining frequency range of signal to be transmitted in the line and transmitting signal in the corresponding line according to the determined frequency range. This mainly means reducing or turning off the transmitting of the signal at a high frequency band and only using low frequency band for data transmitting in the case that the line parameter exceeds the predetermined threshold, so that crosstalk to neighboring lines is reduced. The present invention can effectively overcome the crosstalk against neighboring lines during the channel discovery phase of digital subscriber line and the transfer performance of the digital subscriber line in operation can be guaranteed. The present invention is realized based on the established standard, so the present invention is simple and easy to implement.

Abstract: A method includes transmitting a first signal over a network from a first communication link to a second communication link. The method further includes receiving a second signal with the first communication link from the second communication link. The method further includes canceling signal echo from the first signal present in the second signal with a digital echo canceller. The method further includes providing correction data from a memory array to the digital echo canceller during the cancellation of the signal echo. An associated apparatus is also disclosed.

Abstract: Disclosed is a non-linear echo canceller and method for cancelling echo during full duplex communication in a hands free communication system. An input signal from a far-end talker and an input signal from the output from an echo canceller are received. K spectral subbands are created for each input signal. The spectral echo residual power at each subband is estimated and compared to a clean signal power to calculate a signal to echo ratio. Gains are calculated based on each calculated ratio and non-linear echo is cancelled based on the calculated gains.

Abstract: A method and apparatus for providing echo cancellation are disclosed. For example, the method receives via an echo canceller a first audio signal directed towards an endpoint device, and adds an inserted signal to the first audio signal directed towards the endpoint device. The method determines if a second audio signal being received from the endpoint device comprises the inserted signal, and cancels at least a portion of the second audio signal that is associated with the first audio signal, if the inserted signal is detected.

Abstract: A computer implemented method includes determining crosstalk characteristics of a plurality of digital subscriber lines. The computer implemented method includes identifying pairs of digital subscriber lines, each of the pairs including a first digital subscriber line and a second digital subscriber line, where the first digital subscriber line has a crosstalk characteristic that is correlated to the second digital subscriber line. The computer implemented method also includes rearranging terminations at an access node to assign at least one of the identified pairs of digital subscriber lines to a common vectoring group.

Abstract: A method (200), and electronic communication device (100), for automatically selectively changing from one mode of operation to a speakerphone mode of operation after a call on the electronic communication device (100) is established. The method (200), and electronic communication device (100), perform establishing a call (210) using a first mode of operation that is a mode of operation other than the speakerphone mode of operation. Then there is performed changing automatically (235) from the first mode of operation to the speakerphone mode of operation when a signal amplitude provided by a microphone used in the first mode of operation does not reach a threshold level before an expiration of a predetermined time period.

Abstract: A signal processing system which discriminates between voice signals and data signals modulated by a voiceband carrier. The signal processing system includes a voice exchange, a data exchange and a call discriminator. The voice exchange is capable of exchanging voice signals between a switched circuit network and a packet based network. The signal processing system also includes a data exchange capable of exchanging data signals modulated by a voiceband carrier on the switched circuit network with unmodulated data signal packets on the packet based network. The data exchange is performed by demodulating data signals from the switched circuit network for transmission on the packet based network, and modulating data signal packets from the packet based network for transmission on the switched circuit network. The call discriminator is used to selectively enable the voice exchange and data exchange.

Abstract: A method of acoustic echo cancellation in the VoIP terminal using processing of the far-end signal with the digital adaptive filter in order to obtain the echo estimate that is subtracted from the microphone signal in which the far-end signal, before it is converted to the analog form and passed to the loudspeaker (4), is marked by embedding an encoded digital signature obtained from the signature generator (14) and then detection of the digital signature is performed in the signal collected by the microphone (7) and converted to digital form, depending on the result of the digital signature detection, adaptation of the digital adaptive filter (9) is resumed or stopped.

Abstract: An access node of a communication system comprises a plurality of transmitters adapted for communication with at least one receiver over a plurality of channels. The access node is operative to obtain estimated crosstalk coefficients between a joining channel and an active channel, and to set a power level of at least one signal transmitted over the joining channel based on the estimated crosstalk coefficients. The access node obtains the estimated crosstalk coefficients by first obtaining a subset of the estimated crosstalk coefficients and subsequently determining additional ones of the estimated crosstalk coefficients by applying an interpolation process to the estimated crosstalk coefficients in the subset. The access node sets the power level of the signal transmitted over the joining channel in a manner that ensures maintenance of a desired performance characteristic for the active channel.

Abstract: A method and system in network switch for cancelling multiple sources of network echo, including echo that is combined with a persistent, non-echo signal. A first echo canceller cancels a first echo signal that arises from an echo path in the network switch, and second echo canceller cancels a second echo signal that arises from an echo path in an external network device. The first echo canceller operates by first removing any persistent, non-echo signal that may be combined with the first echo signal, then training on the remaining, pure echo signal. The second echo canceller includes two sub-component echo cancellers that are coordinated so as to provide a minimum error signal for cancellation applied to the second echo signal.

Abstract: Systems and methods are described for selecting tones for far-end crosstalk (FEXT) mitigation. In particular, systems and methods are described for performing far-end crosstalk (FEXT) mitigation within a digital subscriber line (xDSL) system. One embodiment is a method that comprises determining a first set of function values relating to a per-tone signal-to-noise ratio (SNR) for a user among a plurality of users for a number of tones, wherein the number of tones is less than or equal to a total number of tones available for transmission. The method further comprises estimating coefficients for performing FEXT mitigation for the number of tones, performing FEXT mitigation on the number of tones based on the estimated coefficients, and performing FEXT mitigation on a selected group of tones from among the number of tones, wherein the selected group of tones is less than or equal to the number tones.

Abstract: An adaptive feedback estimation and cancellation (AFEC) apparatus includes: a controller for generating and outputting control information by using a synchronization signal from an external synchronization acquisition unit and base station information, in order to remove a feedback signal that exists in a forward/reverse repeater signal to be repeated and then send the forward/reverse repeater signal; a first feedback prediction canceller for adaptively removing a feedback signal that exists in the forward repeater signal based on the control information from the controller and automatically adjusting the gain of the forward repeater signal; and a second feedback prediction canceller for adaptively removing a feedback signal that exists in the reverse repeater signal based on the control information from the controller and automatically controlling the gain of the reverse repeater signal.

Abstract: Apparatus, methods, and machine-readable articles of manufacture enable a telephone system to distinguish between actual events and echoes. Events may include conferees' voice and DTMF commands. In selected embodiments, the system categorizes an event received on one channel as a real event if the energy of the event's waveform is greater than concurrent energy of every other channel of the same conference. Otherwise, the event is stored. When the system receives a new event of the same type on a channel of the same conference during a predetermined period, it compares the energies of the new and stored events. If the energy of the new event exceeds that of the stored event, the new event becomes the stored event, and the old stored event is ignored. At the end of the period, the stored event is categorized as a real event and forwarded to appropriate application code.

Abstract: An apparatus comprising a transmitter configured to transmit a digital subscriber line (DSL) signal in a line from a plurality of subscriber lines according to a reference virtual noise power spectral density (PSD), wherein the reference virtual noise PSD is based on an upper bound for a worst-case far-end crosstalk (FEXT) PSD in the subscriber lines. Also included is an apparatus comprising at least one processor configured to implement a method comprising obtaining an upper bound for a worst-case FEXT PSD for a plurality of subscriber lines, obtaining a referred virtual noise PSD based on the worst-case FEXT PSD for the lines, and determining a bit-loading for the lines based on the referred virtual noise or a scaled version of the receiver-referred virtual noise.

Abstract: A howling detector is provided which can discriminate between howling and a signal having a strong narrow-band component, thereby detecting howling with higher accuracy. The howling analyzer includes a frequency analyzing section for analyzing a frequency of a time signal, a level calculating section for calculating a level of a signal output from the frequency analyzing section, a howling detecting section for deciding whether howling occurs or not by analyzing the level having been calculated by the level calculating section, a periodic signal detecting section for deciding whether or not time progression of the level having been calculated by the level calculating section has periodicity, and a howling deciding section for finally deciding whether howling occurs or not based on decision results of the howling detecting section and the periodic signal detecting section.

Abstract: The coefficient generating section receives a first signal which is the output signal of the microphone of a signal generated by subtracting the output signal of a linear echo canceller from the output signal of the microphone and a second signal which is the output signal of the linear echo canceller. The coefficient generating section detects the minimum value of the variation with time of the ratio of the amplitude of the first signal to that of the second signal and outputs the value of constant times the detected minimum value as a crosstalk coefficient indicating the degree of crosstalk of the echo. The converting section corrects the first signal according to the crosstalk coefficient and the second signal to generate a near-end signal which is the resultant signal of when the echo is removed from the first signal and outputs the near-end signal to an output terminal.

Abstract: An echo signal processing apparatus is disclosed. The echo signal processing apparatus is utilized for generating a cancellation signal by using group phenomenon of a frequency response of an echo signal to remove the echo signal. The echo signal processing apparatus has lower cost and is able to remove the echo efficiently.

Abstract: A method and a device are provided for data processing, including the steps of transmitting a first signal from a first instance to a second instance and sending a second signal from a third instance to the first instance, wherein the second signal corresponds at least partially to the first signal as received at the third instance.

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 system may include a processor and memory. The processor may be configured to perform calibration measurements on the speaker even when the speaker is being used to conduct a live conversation. The processor may be configured to: provide a live output signal for transmission from a speaker; receive an input signal corresponding to the output signal; compute a midrange sensitivity and a lowpass sensitivity for a transfer function derived from a spectrum of the input signal and a spectrum of the output signal; subtract the midrange sensitivity from the lowpass sensitivity to obtain a speaker-related sensitivity; perform an iterative search for current parameters of a speaker model using the input signal spectrum, the output signal spectrum and the speaker-related sensitivity; and update averages of the speaker model parameters using the current parameter values. The parameter averages may be used to perform echo cancellation.

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

Abstract: An adaptive interference cancellation system is described. In one example the system operates by receiving a data signal using a DSL (Digital Subscriber Line) and receiving a reference signal, the reference signal corresponding, in part, to noise on the data signal. The reference signal is classified and a noise cancellation signal is applied to the data signal based on the classification.

Abstract: A system includes a loudspeaker, a microphone, an echo canceling circuit, and an accelerometer coupled to the loudspeaker for providing a reference signal, a(n), to the echo canceling circuit. Preferably, the accelerometer is attached to the loudspeaker. A speaker signal, x(n), is used to drive the loudspeaker and the system also includes a switch for coupling either a(n) or x(n) to the echo canceling circuit, depending upon the volume or loudness of the sounds involved.

Abstract: An apparatus to enable half-duplexing capabilities in a two-way communication device is disclosed. The apparatus estimates the signal power and background noise of a first input signal and a second input signal during approximately the same period. The apparatus further provides at least one control signal based on the result of one or more determinations. These determinations may include whether the estimated signal power of at least one of the first and second input signals exceeds a threshold value; whether the estimated signal power of the first input signal exceeds the sum of a first threshold value and the estimated background noise of the first input signal; and whether the estimated signal power of the second input signal exceeds the sum of a second threshold value and the estimated background noise of the second input signal. Other embodiments for use with two-way communication devices and related methods are also disclosed.

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: Processing requirements for echo cancellation in voice communications are significant and are even more so as the bandwidth of the communication increases. Whilst voice communication occupies a relatively narrow band of frequencies the processing requirements and so forth for wideband communication render echo cancellation difficult to achieve in a cost effective manner. The invention provides for provides echo cancellation within wideband communications by dividing the communications into sub-bands and applying echo cancellation to some sub-bands whilst processing other sub-bands according to the status of the communications. Additional sub-bands are transmitted at either full-duplex or half-duplex.

Abstract: A system and method for adaptive estimation and compensation of clock drift in echo cancellers is provided. The invention includes an acoustic echo cancellation system with a built in adaptive clock drift compensation system. The acoustic echo cancellation system has an AEC component that performs acoustic echo cancellation on data from a capture buffer, by also using information derived from a render buffer. The clock drift compensation system has access to this capture buffer and render buffer. The clock drift compensation system includes a clock drift compensator that calculates, based on the current location of the capture data being processed by the AEC component as well as additional information, the ideal location in the render buffer from which the AEC component should process data. The clock drift compensator further adjusts the current location in the render buffer from which the AEC component processes data based, at least in part, upon this ideal location.

Abstract: The invention concerns a method and device for processing echo between at least two communication devices in order to attenuate, in a picked-up signal of a communication device comprising at least one microphone, the components of the signal broadcasted by at least one other communication device comprising at least one loudspeaker, characterised in that the echo processing device comprises: means for receiving, by means of a connection with at least one other device, information representing at least one broadcasted signal of at least one other communication device, means for modifying the picked-up signal of the communication device according to information representing the broadcasted signal and information representing the coupling separating a loudspeaker of the said at least one other communication device from the microphone of the communication device.

Abstract: A system for canceling acoustic feedback includes an input for receiving a digital audio signal and a processor configured to detect acoustic feedback signal in the digital audio signal and to determine the frequency of the feedback signal The system also includes a plurality of bandpass filters for attenuating the feedback signal. The processor is further configured to: select a bandpass filter from among the plurality of bandpass filters. The selected bandpass filter comprises a response characteristic that attenuates parts of the signal at the frequency of acoustic feedback signal.

Abstract: The present invention relates to a vectoring chip unit in a DSL system connectable to a plurality of DSL ports of a DSL chip comprising means for receiving data from each of the plurality of DSL ports, means for performing pre-coding/cancellation in order to reduce crosstalk and means for transmitting a result of the precoding/cancellation to the respective DSL port. Preferably, a system is provided comprising a plurality of said vectoring chips, whereby each vectoring chip is adapted to handle pre-coding/cancellation of a predetermined frequency range and the frequency ranges of each vectoring chip are non-overlapping.

Abstract: A voice communicating apparatus in which a far-end voice of high sound quality can be outputted from a loudspeaker and an echo which is caused by the outputted far-end voice can be accurately removed with high precision. An analog far-end voice signal supplied from a telephone unit is directly outputted from the loudspeaker and a digital detection sound signal of the far-end voice and a near-end voice detected by a microphone is delayed by a predetermined time and supplied to an arithmetic operating part.

Abstract: An intermediary communication system, the intermediary communication including: (a) a first network interface, configured for transmitting over a first network connection to a first remote end unit a first sound sequence; and for receiving from the first remote end unit a returning sound sequence that is responsive to the first sound sequence; (b) a processor, configured to determine an echo reduction parameter in response to a relationship between a first sound sequence parameter and a returning sound sequence parameter; and (c) a second network interface, for transmitting to a second remote end unit, over a second network connection, a processed sound sequence that was generated in response to the echo reduction parameter from a preprocessed sound sequence which was generated by the first remote end unit.

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 for canceling crosstalk, the method includes the steps of: filtering the signal corresponding to a far end crosstalk source in the frequency domain according to adaptive filter parameter; combining the received signal corresponding to a far end sending signal and the filtered signal in frequency domain corresponding to the far end crosstalk source. A device for canceling crosstalk and a signal processing system.

Abstract: The invention concerns a method, a network element and a computer program product for monitoring the quality of a real-time communication (9) in a packet network (1) comprising at least two test points (3, 4). A packet stream pattern (22) is accessed and injected at a first test point (3). After transmission of the packet stream pattern (22) through the packet network (1), a degraded packet stream pattern (23) is received at a second test point (4). Characteristics of the degraded packet stream pattern (23) are determined and compared with corresponding characteristics of the packet stream pattern (22) injected at the first test point (3). Based on the comparison, data concerning quality degradation caused by transmission over the packet network (1) is evaluated.

Abstract: A voice processing apparatus includes a band dividing portion dividing a first voice signal generated by a first microphone and a second voice signal generated by a second microphone into predetermined frequency bands, a sound source segregating portion segregating an echo component of a voice emitted by a first sound source included in a voice emitted by a second sound source in each of the predetermined frequency bands based on the power of the first and second microphones, and a band synthesis portion synthesizing the first and second voice signals from which the echo component of the first sound source has been segregated by the sound source segregating portion into a voice signal including the voice emitted by the first sound source and a voice signal including the echo component of the first sound source.

Abstract: Far-end audio signal data generated by a far-end device during voice cross-talk between the far-end device and a near-end device is received via a network. Energy levels of frequency components of the received far-end audio signal data are determined. Near-end audio signal data produced by a microphone array is received, the near-end audio signal data including components contributed by a loudspeaker playing at least some of the far-end audio signal data. Which frequency components of the near-end audio signal data to use to compute likely near-end sound source directions is controlled dynamically, where the controlling is based on the determined energy levels of the frequency components of the received far-end audio signal data.

Abstract: A method and system for xDSL crosstalk cancellation is provided. The method includes dividing xDSL signals into a plurality of signal sets; and connecting signals from a same signal set to a same processing unit to be processed. A digital subscriber line access multiplexer (DSLAM) includes a line switching control module and at least one processing unit.

Abstract: An input sound processor compares power at each frequency component of an input sound with a reference value, and sets multiplication points indicating frequency components at which the total power of the input sound is to be determined. A product-sum operation is performed at the multiplication points on the power at each frequency component and the square amplitude of each filter coefficient indicating the transfer characteristic from a loudspeaker to a microphone to estimate the total power of the input sound at the position of the microphone.

Abstract: An echo canceller which can respond to a sudden change in echo characteristics in real time and does not require an alteration outside the echo canceller, includes a smoothed sending-speech signal calculation means for calculating a smoothed sending-speech signal from the sending-speech signal, the smoothed sending-speech signal being obtained by smoothing the sending-speech signal; a smoothed receiving-speech signal calculation means for calculating a smoothed receiving-speech signal from the receiving-speech signal, the smoothed receiving-speech signal being obtained by smoothing the receiving-speech signal; a delay time information generation means for obtaining delay time information reflecting delay characteristics of an echo path in accordance with a correlation between the smoothed sending-speech signal and the smoothed receiving-speech signal; and an update information generation means for obtaining update information indicating execution or suspension of updating of the tap coefficients of the adapti

Abstract: The present invention relates to detection of echo in telecommunications networks. The invention provides a method of echo detection comprising the steps of: generating a series of inbound vectors for an inbound signal; generating a series of outbound vectors for an outbound signal; repeating a predetermined number of comparison steps comprising the sub-steps of selecting an outbound vector from the outbound vectors; selecting an inbound vector from the inbound vectors; comparing said outbound vector with said inbound vector and with successive inbound vectors to generate a plurality of similarity metrics; and determining a relative position of the compared outbound vector having maximum correlation with said inbound vector; and counting the number of times each relative position is determined to be the position of maximum correlation.

Abstract: An echo cancellation system for use with a controller area network having at least two physical layers is disclosed. The system includes a flip flop circuit and a first resistor capacitor tank circuit and a second resistor capacitor tank circuit. The echo cancellation system uses a transition dependent delay in the circuit that will be tuned to be longer than the propagation delay of the CAN transceivers but less than the bit sample point for the CAN network thus eliminating any possibility of oscillation occurring within the CAN network during transfer of information between physical layers.