Abstract: In some embodiments, a method receives a first signal that is sent in a first direction in a network. Communications in the network are full duplex communications in a same frequency band. The first signal is amplified in the first direction. The method trains a first echo canceller to cancel a first echo signal from the first signal where the first echo signal is received in a second direction. After training the first echo canceller, the trained first echo canceller is enabled. The method receives a second signal in the second direction that is sent in the second direction in the network. The second signal is amplified in the second direction. The method trains a second echo canceller to cancel a second echo signal received in the first direction from the second signal where the first echo canceller cancels the first echo signal that is received in the second direction.

Abstract: Systems and methods are provided for acoustic echo canceling. In one embodiment, a system for canceling acoustic echoes comprises one or more microphones configured to pick up sound generated by a sound source and transferred from the sound source to the one or more microphones via one or more unknown transfer paths having one or more unknown transfer functions, and to provide one or more electrical microphone signals therefrom, one or more adaptive filters for approximating the one or more unknown transfer functions with one or more estimated transfer functions and filtering one or more electrical signals representative of the sound generated by the sound source with the one or more estimated transfer functions to provide one or more estimated signals therefrom, and a microphone calibration block configured to individually attenuate or amplify the one or more electrical microphone signals dependent on the one or more estimated transfer functions.

Abstract: A Micro-Electro-Mechanical System (MEMS) circuit and a method for reconstructing an interference variable are provided. The MEMS circuit includes a MEMS device configured to generate a MEMS signal; a control circuit configured to detect a switched-on state or switched-off state of at least one device and configured to generate a control signal at least partly depending on the switched-on state or the switched-off state; a reconstruction filter configured to determine an interference signal that is partly generated by the at least one device, using the generated control signal; and a subtractor configured to subtract the determined interference signal from the MEMS signal.

Abstract: Method and apparatus for reducing distortion echo are provided. K-path amplification and pre-distortion process are performed to the downlink reference signal to obtain K-path pre-distorted signals. Afterwards, filtering is performed using the self-adaptive filters which correspond to the downlink reference signal x(t) and the K-path pre-distorted signals to obtain the filtering signals. Error signals are obtained by calculating differences between the target signal and each of the filtering signals. The minimum-value fusion process is performed to the error signals to obtain the residual signal which is then output as the final self-adaptive echo cancellation. In embodiments of the present disclosure, the residual signal is relatively small as the minimum-value fusion process is performed to the error signals. That is to say, echo loss is relatively great. Therefore, the method may provide echo loss with high amplitude under a situation that a speaker has relatively serious distortion.

Abstract: The invention relates to a method and to a device for reducing interference between a first and a second digital subscriber line, DSL. A corresponding communication system is also provided. The invention provides a method wherein interference between a first and a second digital subscriber line is reduced, the method comprising the steps of determining line characteristics of the second DSL and reducing a transmission power of the first DSL based on the line characteristics of the second DSL.

Abstract: An audio-based system may perform noise and echo suppression by initially processing an audio signal that is subject to acoustic echo or echo resulting from other system characteristics. The audio signal is processed in the time domain using an adaptive echo-cancellation filter. The audio is then further processed in the frequency domain to simultaneously reduce background noise and residual echo.

Abstract: A personal audio device, such as a wireless telephone, includes noise canceling circuit that adaptively generates an anti-noise signal from a reference microphone signal and injects the anti-noise signal into the speaker or other transducer output to cause cancellation of ambient audio sounds. An error microphone may also be provided proximate the speaker to measure the output of the transducer in order to control the adaptation of the anti-noise signal and to estimate an electro-acoustical path from the noise canceling circuit through the transducer. A processing circuit that performs the adaptive noise canceling (ANC) function also either adjusts the frequency response of the anti-noise signal with respect to the reference microphone signal, and/or by adjusting the response of the adaptive filter independent of the adaptation provided by the reference microphone signal.

Abstract: A method for detecting a double-talk condition and an echo cancellation system using the same are introduced herein. According to an exemplary embodiment, the method may be adapted to an double-talk detector of an echo cancellation system and includes the following steps: obtaining an input signal with a first power; computing a second power, wherein the second power is a power combination of at least one estimated signal; determining whether the double-talk condition occurs according to a relationship between the first power and the second power, when the first power is larger than the second power, the double-talk condition occurs.

Abstract: In one embodiment, an echo canceller configured to cancel echo in a wideband voice conference is provided. A double-talk condition may be when a plurality of users are speaking substantially simultaneously. When a double-talk condition is detected in the wideband conference, a high-frequency process is enabled and used to process signals in the high band to reduce echo. Accordingly, echo in the high band may not be produced by end devices being used by the users' speaking. Also, the users speaking have the echo cancelled in the low band and substantial echo does not result. This results in the users speaking experiencing the conference in the narrowband. The other users that are not speaking, however, continue to receive wideband signals. The users not speaking also continue to have echo cancellation performed for the high band and low band because these users are not speaking and thus attenuation of their voices is not a consideration.

Abstract: An echo canceller for executing adaptive processing for canceling an echo component mixed with an audio input signal includes a volume ratio learner configured to compute a volume ratio between an audio output signal externally outputted and the audio input signal mixed with an echo component caused by reflection of the audio output signal to the audio input signal, thereby learning the volume ratio in a regular status in own apparatus, a double-talk detector configured to detect the double-talk status depending on whether a this-time volume ratio computed this time adapts to a double-talk status predicted by the learning of volume ratio and an echo cancel processor configured to control a learning operation of the echo component for the adaptive processing on the basis of a result of the double-talk status detection by the double-talk detector.

Abstract: An audio communication system includes a first analog/digital convertor, a second analog/digital convertor, and an echo canceller. The first analog/digital convertor converts an audio signal, which is input from a microphone, to a data signal. The second analog/digital convertor that converts an audio signal, which is input from an auxiliary input terminal, to a data signal. When an audio signal transmitted from the outside is received and output from a speaker, the echo canceller prevents an output of the speaker from being input through the microphone and echoed back to a recipient or remote device. An operating frequency of the echo canceller is relatively lower than a sampling frequency of the second analog/digital convertor.

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 includes an echo canceller having an audio signal input and an audio signal output and dynamic pre-conditioning logic. The dynamic pre-conditioning logic is operatively coupled to the echo canceller audio signal output as a feedback signal and has a dynamic pre-conditioning logic output operatively coupled to the echo canceller audio signal input. The dynamic pre-conditioning logic is also operative to receive an audio signal input from at least one microphone. The dynamic pre-conditioning logic is operative to analyze the feedback signal to obtain at least one characteristic, and pre-condition the audio signal input, based on the at least one characteristic of the feedback signal, and provide a pre-conditioned audio signal at the echo canceller audio signal input. The echo canceller audio signal output is then provided to a noise suppressor for the send path of a full duplex communication channel.

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: A Voice Echo Cancellation system and process are disclosed. According to one embodiment, the present invention comprises an input audio source that is converted to a digital signal via a PCM convertor where the amplitude of the input audio source is sampled at regular intervals and translated into digital PCM audio. The resulting PCM audio is analyzed for its dynamic and harmonic content. Corrective sound waves are then generated that reduce the repetition of the original audio (echos). The corrective sound waves are applied to the repeating audio echoes to create corrected sound waves. The corrected audio is then outputted from the system.

Abstract: An adaptive filter unit outputs a send-mid signal obtained by eliminating echo from a send-in signal, and a power comparing unit calculates a power ratio between received signal power and send-mid signal power. When a receiver ST detecting unit detects a single talk state at a receiving side, an acoustic coupling amount estimating unit estimates and updates the estimated amount of acoustic coupling from the power ratio. A residual echo power estimating unit estimates estimated residual echo power from the received signal power and the estimated amount of acoustic coupling, and a signal-to-echo ratio estimating unit estimates a ratio between the send-mid signal power and the estimated residual echo power. An amplitude suppression coefficient determining unit determines the amplitude suppression coefficient corresponding to the ratio, and an amplitude suppression unit amplitude suppresses the send-mid signal.

Abstract: Methods and apparatus are provided for acoustic echo cancellation in a speech signal. Acoustic echo is cancelled by inserting at least one tone in the speech signal, wherein the at least one tone is substantially inaudible to a listener; determining a clock skew between two sampling clocks based on a frequency shift of the at least one tone; re-sampling the speech signal based on the determined clock skew; and performing the acoustic echo cancellation using the re-sampled speech signal. The provided acoustic echo cancellers can be implemented, for example, as terminal-based and/or network-based acoustic echo cancellers. The tone optionally comprises an inaudible tone or multiple tones. The tone generation can be limited to only when a speech power in the vicinity of the tone frequency is larger than a pre-determined threshold, or to the beginning of a call. A level of the tone can optionally be controlled so that the tone is masked by the speech signal.

Abstract: An embodiment of an apparatus for computing control information for a suppression filter for filtering a second audio signal to suppress an echo based on a first audio signal includes a computer having a value determiner for determining at least one energy-related value for a band-pass signal of at least two temporally successive data blocks of at least one signal of a group of signals. The computer further includes a mean value determiner for determining at least one mean value of the at least one determined energy-related value for the band-pass signal. The computer further includes a modifier for modifying the at least one energy-related value for the band-pass signal on the basis of the determined mean value for the band-pass signal. The computer further includes a control information computer for computing the control information for the suppression filter on the basis of the at least one modified energy-related value.

Abstract: A method and apparatus for canceling an echo in audio communication is disclosed. The method comprises receiving an audio signal from a network and subsequently detecting a mixture audio signal comprising a target audio signal and an echo audio signal, the echo signal corresponding to the received audio signal. The method then comprises estimating the target audio signal by determining magnitude spectrograms for the mixture and received audio signals respectively, estimating a magnitude spectrogram of the target audio signal dependent on those of the mixture and received audio signal, and generating an output audio signal that estimates the target audio signal, the output audio signal being dependent on the estimated magnitude spectrogram.

Abstract: An input signal is supplied to a loudspeaker-room-microphone system having a transfer function and that provides an output signal. An adaptive filter unit models the transfer function of the loudspeaker-room-microphone system and provides an approximated output signal, where the output signal and the approximated output signal are subtracted from each other to provide an error signal.

Abstract: A system and method are presented for acoustic echo cancellation. The echo canceller performs reduction of acoustic and hybrid echoes which may arise in a situation such as a long-distance conference call with multiple speakers in varying environments, for example. Echo cancellation, in at least one embodiment, may be based on similarity measurement, statistical determination of echo cancellation parameters from historical values, frequency domain operation, double talk detection, packet loss detection, signal detection, and noise subtraction.

Abstract: The present disclosure relates to a signal analyzer for processing an overlapped input signal frame comprising 2N subsequent input signal values. The signal analyzer comprises: a windower adapted to window the overlapped input signal frame to obtain a windowed signal, wherein the windower is adapted to zero M+N/2 subsequent input signal values of the overlapped input signal frame, wherein M is equal or greater than 1 and smaller than N/2; and a transformer adapted to transform the remaining 3N/2?M subsequent windowed signal values of the windowed signal using N?M sets of transform parameters to obtain a transformed-domain signal comprising N?M transformed-domain signal values.

Abstract: A new acoustic echo suppressor and method for acoustic echo suppression is described herein. Exemplary embodiments of the acoustic echo suppressor use one linear regression model for each subband. The linear regression model for each subband may operate on the squared magnitude of the input samples as well as corresponding cross-products. In this way, accurate and robust estimates of the echo signal in each subband can be obtained, thereby providing good echo reduction while keeping the signal distortion low.

Abstract: An apparatus is provided for suppressing an echo signal included in a measured signal corresponding to a measured sound. In the apparatus, the measured signal and a reference signal in a time domain are transformed into a frequency domain, and calculated for obtaining each value of a ratio and a correlation between the measured signal and the reference signal in the frequency domain. With executing a comparison of the values of the ratio and the correlation, a coefficient is derived, where a product of the coefficient and the measured sound in the frequency domain gives an estimated value of the echo signal. The echo in the measured signal is suppressed with subtracting the estimation of the echo signal from the measured signal, respectively in the frequency domain.

Abstract: An echo cancelling algorithm in a communication device initializes a step size value used in an adaptive echo filter based on a background noise signal power level relative to a power level of a received signal and a power level of an echo estimate relative to an output of an echo canceller. The algorithm then adjusts the step size value. One aspect adjusts the step size based on the detection of large fast fourier transform values at one, or more, disturbing-signal frequencies. Another aspect estimates residual echo energy to adjust an estimated echo energy, which then is used to set a double talk flag if a transmit signal has much more power than the estimated echo signal. Another aspect compares transmit signal power to a decimated version of the transmit signal power and sets the double talk flag if the former exceeds the latter by a predetermined amount.

Abstract: A system and methods for pre-configuring echo cancellers are shown and described. The system includes a storage device for storing one or more settings of the echo cancellation parameters developed during one or more previously established data connections, and an echo canceller for cancelling echo associated with one or more data connections according to echo cancellation parameters, the echo canceller initially configuring the echo cancellation parameters according to settings developed during previous data connections.

Abstract: To assist with the detection of unfiltered device(s), a system observes how the received noise changes between two links as the transmit signal is changed. Harmful unfiltered nonlinear devices will generate significant noise that depends on the transmitted signals; therefore, this additional noise can be quantified to some extent by comparing the observed noise for two different transmit signals. The total noise can be determined from the SNR if the received signal is known. The received signal may be read directly in some non-standard systems, or it may be determined from the known transmit signal and channel attenuation, which is sometimes the case in standard-compliant links, but often with a relatively large error. To circumvent this problem, certain embodiments of this invention only consider the change in noise between two links with the same channel attenuation. This differential comparison makes it unnecessary to accurately know the channel attenuation.

Abstract: A communications device that is configured to detect double talk is described. An echo canceller is configured to cancel an echo from an input signal using an adaptive filter. A double-talk detector provides a double-talk statistic. The double-talk statistic is proportional to the ratio of the remaining echo energy in the cancellation error signal and the total cancellation error energy.

Abstract: Methods and techniques to implement a digital signal processor for avoidance of audio feedback are disclosed, in particular, audio signal processing systems that reduce the requirement for physical segregation of sound acquisition and diffusion zones. In a more general aspect, the components and techniques described herein provide a for a sound space and sound processing equipment such that sound travelling electronically in a loop through the sound processing equipment that is output into a physical sound diffusion zone, received at the input to the sound processing equipment, and then re-amplified, etc. is attenuated over that loop by frequency modification. The frequency modification is such that, at least for some signals, on each pass through the loop, the sound processing equipment will attenuate or amplify individual sub-bands of the frequency spectrum of the audio signal that is received at the input of the sound processing equipment.

Abstract: A howling suppression device that can reduce quality deterioration of processed sound includes: a delay unit delaying the input signal to output the delayed input signal as a reference signal; a signal separation unit including an adaptive filter extracting a periodic signal component from the reference signal by adaptively updating a filter coefficient; a howling detection unit detecting an occurrence of howling using at least a signal of the periodic signal component output from the adaptive filter; and a howling suppression unit. The howling suppression unit includes: a suppression filter obtaining the updated filter coefficient from the adaptive filter with timing when the howling detection unit detects the occurrence of the howling, to extract the periodic signal component from the reference signal based on the filter coefficient; and a subtractor subtracting the periodic signal component from the input signal so as to output a signal obtained by the subtraction.

Abstract: A preferred embodiment of an apparatus for computing filter coefficients for an adaptive filter for filtering a microphone signal so as to suppress an echo due to a loudspeaker signal includes an extractor for extracting a stationary component signal or a non-stationary component signal from the loudspeaker signal or from a signal derived from the loudspeaker signal, and a computer for computing the filter coefficients for the adaptive filter on the basis of the extracted stationary component signal or the extracted non-stationary component signal.

Abstract: An apparatus may include a noise estimation unit configured to determine a noise spectrum associated with a noise signal, and generate a filter based on the determined noise spectrum. The apparatus may also include a noise synthesis unit configured to generate a colored noise using the filter generated by the noise estimation unit. The apparatus may be incorporated in an echo canceller.

Abstract: Conferencing products for conducting a conference between a local and two or more distant parties through an electronic channel, wherein distant party audio is routed to the distant parties to provide full audio. These conferencing products are capable of open-air interaction with participants in the locality of a product. Acoustic echo cancellation is applied, which may be accomplished through a single set of coefficients or by a plurality of coefficients and/or echo cancelers on different incoming audio streams and speaker combinations. Privacy modes are provided, whereby one distant party cannot hear the communications of another distant party; those modes including a coaching mode where the privacy is one-way and a mediation mode where the privacy is two-way. Detailed information on various example embodiments of the inventions are provided in the Detailed Description below, and the inventions are defined by the appended claims.

Abstract: Methods and apparatus are provided for acoustic echo cancellation in a speech signal. Acoustic echo is cancelled by inserting at least one tone in the speech signal, wherein the at least one tone is substantially inaudible to a listener; determining a clock skew between two sampling clocks based on a frequency shift of the at least one tone; re-sampling the speech signal based on the determined clock skew; and performing the acoustic echo cancellation using the re-sampled speech signal. The provided acoustic echo cancellers can be implemented, for example, as terminal-based and/or network-based acoustic echo cancellers. The tone optionally comprises an inaudible tone or multiple tones. The tone generation can be limited to only when a speech power in the vicinity of the tone frequency is larger than a pre-determined threshold, or to the beginning of a call. A level of the tone can optionally be controlled so that the tone is masked by the speech signal.

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 method for determining coefficients of a family of cascaded second order Infinite Impulse Response (IIR) parametric filters used for equalizing a room response. The method includes determining parameters of each IIR parametric filter from poles or roots of a reasonably high-order Linear Predictive Coding (LPC) model. The LPC model is able to accurately model the low-frequency room response modes providing better equalization of loudspeaker and room acoustics, particularly at the low frequencies. Advantages of the method include fast and efficient computation of the LPC model using a Levinson-Durbin recursion to solve the normal equations that arise from the least squares formulation. Due to possible band interactions between the cascaded IIR parametric filters, the method further includes optimizing the Q value of each filter to better equalize the room response.

Abstract: A communication apparatus capable of echo cancellation is provided. The apparatus is configured to be provided with an input signal receiving interference. The input signal has a frequency range. The interference includes an echo component and a noise component. The apparatus has a first interference eliminator configured to reduce an echo component included in the input signal so as to produce an intermediate signal. The apparatus has a selector configured to select one of the input signal and the intermediate signal. The apparatus has a second interference eliminator configured to reduce at least one of a noise component and an echo component included in one of the input signal and the intermediate signal selected by the selector.

Abstract: This disclosure describes a personal speakerphone device for teleconferencing a local participant with near side audio data and a far side participant with far side audio data. The device includes a controller that couples to a speaker and a microphone inside an enclosure that couples the speaker and the microphone within a coupling frequency range, and where the enclosure houses the speaker, the microphone, the controller, analog to digital conversion circuitry, digital to analog conversion circuitry, supporting circuitry, and one or more ports for data through which audio data is transferred. Additionally, the device includes a first frequency band decomposer, a second frequency band decomposer, one or more echo cancellers, and a converger. Further, the device includes a primary doubletalk detector and a secondary doubletalk detector.

Abstract: A method and system for clear signal capture comprehend several individual aspects that address specific problems in improved ways. In addition, the method and system also comprehend a hands-free implementation that is a practical solution to a very complex problem. Individual aspects comprehended related to echo and noise reduction, and divergence control.

Abstract: In an echo processing method and device which can detect an accurate echo section without effects of a far end signal, an echo delay, and a reduction of an echo cancellation amount, a signal of a specified frequency band is generated in conformity with a near end signal, and the signal of the specified frequency band is added to the near end signal to form a transmitting signal. Receiving signals are separated into the signal of the specified frequency band and a signal of a band other than the specified frequency band. An echo section is detected based on the signal of the specified frequency band separated. An echo component in the signal of the band other than the specified frequency band is removed and a level of the echo component is detected based on the near end signal in the echo section.

Abstract: A method and system for determining and compensating for a non-linearity in a hands-free acoustic telecommunication device is disclosed. The method determines a back electromotive force signal induced in a loudspeaker from at least one of a coil voltage, a current signal and estimates of coil resistance and inductance, estimating at least one of a cone position and a cone velocity from the BEMF integrated with respect to time and determining an estimate of an echo value from a series connection of an estimated inverse of a force factor function primitive and the estimated acoustic impulse response; and outputting the estimated echo value.

Abstract: A method and system for clear signal capture comprehend several individual aspects that address specific problems in improved ways. In addition, the method and system also comprehend a hands-free implementation that is a practical solution to a very complex problem. Individual aspects comprehended related to echo and noise reduction, and divergence control.

Abstract: A conferencing unit reduces or eliminates percussive noise in audio that it outputs to a far-end. The unit filters near-end audio into a plurality of bands and calculates instantaneous energies for each band. The energies in first and second ranges of the bands are summed, and a difference between the two sums is compared to a threshold value to determine whether speech is present in the near-end audio received. The first range is preferably a mid-band range of frequencies of 300 to 600-Hz, while the second range is preferably an outer-band range of 100-Hz to 300-Hz and 600-Hz to 14-kHz. Based on the determination, the conferencing unit adjusts an output level of the near-end audio to reduce or eliminate any percussive noise in output audio while there is not a substantial amount of speech in the near-end audio.

Abstract: A communications device is presented for providing bi-directional audio communications between a near-end user and a far-end user via a bidirectional communications channel. The communications device includes an adaptive echo canceller receiving a near-end audio signal and a far-end audio signal and providing an echo-canceled near-end audio signal for transmission to the far-end user via the communications channel. The adaptive echo canceller includes a first bank of analysis filters for filtering the near-end audio signal, a second bank of analysis filters for filtering the far-end audio signal, and a bank of synthesis filters for filtering sub-band echo-canceled signals generated within the adaptive echo canceller. The first and second filter banks have a frequency response optimized to reduce echo residual gain.

Abstract: A technique of echo cancellation in a communication system. A method and/or apparatus of echo cancellation that may be suitable for performing echo cancellation under single talk and double talk conditions. A method and/or apparatus of echo cancellation that may significantly reduce a residual echo in a single talk environment (e.g. present in many telecommunications systems) without distorting a near end signal in a double talk environment. A method and/or apparatus of echo cancellation that may reduce a residual echo in single talk and double talk environments by applying a post-processing technique to an ECLMS algorithm.

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: Disclosed herein are portable teleconferencing products that implement a doubletalk detector in a low frequency range or in a frequency range commensurate with the frequencies through which sound may be efficiently transferred between a speaker and a microphone through an enclosure. Also disclosed herein are teleconferencing systems that implement a secondary doubletalk detector, a non-presumptive doubletalk detector, a confirmatory doubletalk detector, and/or a false doubletalk detector, whereby echo cancellation coefficients may be better adapted after echo path changing events through the use of accelerated coefficient adaptation or half-duplex operation until adaptation is restored.

Abstract: A communication device is provided that includes a speaker for outputting a ringing signal, a microphone for receiving the ringing signal, and logic configured to analyze the spectral content of the received ringing signal and adjust noise reduction parameters and echo cancellation parameters based on the analyzed spectral content of the received ringing signal.

Abstract: There is disclosed a multi-carrier transceiver system for use in echo cancellation. The transceiver system is arranged to generate from input data a multicarrier transmit signal comprising a plurality of multi-carrier data blocks. The transceiver system also has a multi-carrier signal receiver responsive to an input multi-carrier receive signal received from a remote signal transmitter. An echo canceller is arranged to generate from a pair of multi-carrier data blocks (u) a set of frequency-domain echo parameters for use in generating an echo signal, to generate the echo signal using the frequency-domain echo parameters, and to input the echo signal to the multi-carrier signal receiver for use in generating an echo-cancelled receive signal from the input receive signal and the echo signal.

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.