Abstract: The spirit of the present invention is to vary the step size parameter in accordance with the error signal and the output acoustic signal, wherein the filter is easy to implement, nonparametric VSS-NLMS algorithm which employs the mean-square error and the estimated system noise power to control the step-size update. The new nonparametric VSS-NLMS algorithm has been shown to perform with fast convergence rate, good tracking, and low mis-adjustment. In comparison with existing VSS-NLMS algorithms, the proposed algorithm has demonstrated consistently superior performance both in convergence and for final error level relative to published algorithms in application on both simulated data and real speech data.

Abstract: Disclosed is an apparatus and method for cancelling a wideband acoustic echo that controls a data flow of data that is transmitted and received using a central processing unit, and calculates data using a calculation processing unit configured to be distinct from the central processing unit.

Abstract: This disclosure provides a network apparatus with communication ports, each connected to multiple channels, assigned a seed for eliminating the interference among the communication ports, and operable in a master or slave mode, each channel having a communication unit which comprises: an echo canceller, a near-end crosstalk (NEXT) canceller, a decision feedback equalizer, and a control unit configured for controlling the communication ports if the echo canceller and the decision feedback equalizer keep updating their filter coefficients according to the same symbol during a first pre-determined time-interval; wherein when more than two of the communication ports operate in the master mode, the control unit assigns different seeds to the more than two communication ports; and wherein when successive two of the communication ports operate in the slave mode, the control unit stops either the echo canceller or the NEXT canceller from updating their filter coefficients during a second pre-determined time-interval.

Abstract: A method for controlling adaptation of a feedback suppression filter in a hearing aid comprises the step of suspending the adaptation of the feedback suppression filter when the level of a hearing aid reference signal is below a predetermined threshold. The invention further provides a hearing aid with a feedback suppression filter and with means for suspending adaptation of the feedback suppression filter.

Abstract: An audio-based system may perform echo cancellation by decomposing an input audio signal and a corresponding reference signal into sub-signals corresponding to different frequency bands and implementing adaptive filtering independently for each frequency band. Computational resources may be allocated differently for each of the frequency bands depending on observed conditions. Orders of adaptive FIR filters may be varied during operation. Double talk detection and echo path change detection may be implemented for some frequency bands and not others. Filter adaption rates may also be assigned independently to each of the FIR filters, and may be changed during operation.

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: 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: The present invention relates to an echo canceller (300) for estimating a model of an echo signal and a method thereof. The echo canceller comprises at least an adaptive main filter (301) for modeling the echo signal and an adaptive shadow filter (302) for modeling the echo signal. The adaptive main filter is an adaptive filter for which the adaptation speed is proportional to a system noise estimate and the adaptation speed of the shadow filter (302) is faster than the adaptation speed of the adaptive main filter (301). The echo canceller (300) comprises a processor (303) for determining whether the adaptive shadow filter models the echo signal better than the adaptive main filter and an updater (304) for updating the system noise estimate of the adaptive main filter (301) if the adaptive shadow filter models (302) the echo signal better than the adaptive main filter (301).

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: Described herein are multi-channel noise suppression systems and methods that are configured to detect and suppress wind and background noise using at least two spatially separated microphones: at least one primary speech microphone and at least one noise reference microphone. The multi-channel noise suppression systems and methods are configured, in at least one example, to first detect and suppress wind noise in the input speech signal picked up by the primary speech microphone and, potentially, the input speech signal picked up by the noise reference microphone. Following wind noise detection and suppression, the multi-channel noise suppression systems and methods are configured to perform further noise suppression in two stages: a first linear processing stage that includes a blocking matrix and an adaptive noise canceler, followed by a second non-linear processing stage.

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: Multi-channel noise suppression systems and methods are described that omit the traditional delay-and-sum fixed beamformer in devices that include a primary speech microphone and at least one noise reference microphone with the desired speech being in the near-field of the device. The multi-channel noise suppression systems and methods use a blocking matrix (BM) to remove desired speech in the input speech signal received by the noise reference microphone to get a “cleaner” background noise component. Then, an adaptive noise canceler (ANC) is used to remove the background noise in the input speech signal received by the primary speech microphone based on the “cleaner” background noise component to achieve noise suppression. The filters implemented by the BM and ANC are derived using closed-form solutions that require calculation of time-varying statistics of complex frequency domain signals in the noise suppression system.

Abstract: An echo path change detector may be used to control the rate of adaptation in an acoustic echo canceller. When echo path change is declared, the rate of adaptation may be increased. However, echo path change should not be declared in the presence of double talk, because rapid adaptation during double talk is undesirable. Accordingly, various features are disclosed for detecting echo path changes while avoiding the declaration of such changes in the presence of double talk.

Abstract: By using the adaptive filter, the reference input signal is processed so as to identify a pseudo-signal of a particular signal to be deleted. The pseudo-signal is subtracted from the mixture containing a target signal inputted from a microphone, the particular signal to be deleted, and a noise so as to obtain an error signal. A stationary noise is estimated to obtain a stationary noise estimated value. A non-stationary noise is estimated to obtain a non-stationary noise estimated value. The stationary noise estimated value is mixed with the non-stationary estimated value to obtain a mixed noise estimated value. An update amount is calculated according to a correlation value between the error signal and the reference input signal, and the mixed noise estimated value. According to the update amount, a coefficient of the adaptive filter is updated.

Abstract: Example methods, apparatus and articles of manufacture to cancel echo for a communication path having long bulk delays are disclosed. A disclosed example method includes determining a first location of a first coefficient having a largest magnitude of a first plurality of magnitudes associated with a first plurality of coefficients of a first phase; determining a second location of a second coefficient having a largest magnitude of a second plurality of magnitudes associated with a second plurality of coefficients of a second phase different than the first phase; comparing a difference between the first and second locations to a threshold; and, when the difference is less than the threshold, selecting a first offset based on a greater of the magnitude of the first coefficient and the magnitude of the second coefficient; and cancelling an echo contained in a signal using the first offset.

Abstract: An echo prevention circuit comprises an input terminal to which a first input signal is input; a first FIR filter into which the first input signal is input through the input terminal; a second FIR filter into which the first input signal is input at the same time as into the first FIR filter; an input/output terminal to which an output signal of the first FIR filter is output or a second input signal is input; a subtracter that subtracts an output signal of the second FIR filter from a combined signal of the output signal of the first FIR filter and the second input signal; and an output terminal to which an output signal of the subtracter is output. The first and the second FIR filters have such filter coefficients that the output signal through the output terminal has only the output signal from the first FIR filter removed.

Abstract: Methods and devices for improved echo cancellation in high noise environments, such as in-car speakerphones and the like, are provided that improve cancellation convergence in a high noise environment and in the presence of double talk. A new update method enables a robust echo canceller to update with reduced concern for the canceller being in a receive state or a double talk state. If a signal generated by a local talker is substantially the same as high noise, in terms of corruption of the echo canceller, the echo canceller converges in a robust manner even with continuous high noise at the input.

Abstract: A speakerphone having transmit path with a microphone and a first amplifier having a gain determined by a controller and an input coupled to the microphone, a receive path with a second amplifier having a gain determined by the controller and a speaker coupled to an output of the second amplifier, and operable in a half-duplex mode using an adaptive echo canceller to at least partially remove from the transmit path acoustically-coupled signals from the receive path. The controller increases the gain of the first amplifier and decreases the gain of the second amplifier when a level of signals in the transmit path exceed a level of signals in the receive path, and decreases the gain of the first amplifier and increases the gain of the second amplifier when the level of signals in the receive path exceed the level of signals in the transmit path.

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: A method for processing an audio signal is described comprising receiving a first audio signal via a first receiving path comprising a first microphone; receiving a second audio signal via a second receiving path comprising a second microphone; and performing echo suppression of the first audio signal based on the first audio signal and the second audio signal.

Abstract: An audio conferencing system having a base station, a loud speaker and one or more microphones generally operates to receive a F.E. audio signal, play the signal, and to receive a N.E. audio signal at the microphone that includes the F.E. audio signal modified by an echo path and a N.E. audio signal. The base station includes an adaptive filter whose length is controlled to be variable depending upon activity detected at each of one or more filter elements comprising the adaptive filter.

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: Systems and methods are discussed for providing a solution for managing a number of echo cancellers and maintaining their configuration. A software tool is provided to verify and remove incorrect configurations that may result in echo in voice conversations. The configurations are updated using a pre-defined configuration, another echo canceller, or a template. Old configurations for the echo cancellers are stored and may be re-loaded (as a pre-defined configuration) if a problem is encountered with the updated configurations.

Abstract: To provide a noise reduction transmitter which can secure clarity of sounds collected in very noisy environments and maintain a quality of sounds without devising a noise insulation cover particularly. A transmission microphone 7 is arranged inside a noise insulation cover 2 worn on and covering at least a user's 1 mouth. A noise detection microphone 9 which detects external noises is arranged outside the noise insulation cover, and a noise component cancellation circuit 11 is provided which generates a noise component cancellation signal based on an output signal from the noise detection microphone. An electroacoustic transducer 8 is arranged in the noise insulation cover to reproduce a noise component cancellation sound based on an output signal from the noise component cancellation circuit 11.

Abstract: An active noise control (ANC) system may be implemented to both sides of a fan, such that both directions of the noise emitting are treated to reduce the overall noise. The impact on airflow is minimal, and the technique is very effective in a broad range of low frequencies. Passive sound-absorbing materials may be included for attenuation of high frequencies. The resulting quiet fan produces a low level of noise compared to any other device based on fan, which produces the same capacity of airflow. The quiet fan may be incorporated in any mechanical system which requires airflow induction such as: computers, air conditioners, machines, and more.

Abstract: An audio conferencing system having a base station, a loud speaker and one or more microphones generally operates to receive a F.E. audio signal, play the signal, and to receive a N.E. audio signal at the microphone that includes the F.E. audio signal modified by an echo path and a N.E. audio signal. The base station includes an adaptive filter whose length is controlled to be variable depending upon activity detected at each of one or more filter elements comprising the adaptive filter.

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: Echo cancellation is provided in a telephone device by calculating a metric value indicative of a similarity between a representation of an incoming signal from a far end talker and a representation of an outgoing signal from a near end talker. A threshold value is derived based on the metric value. A portion of the outgoing signal is removed by using the threshold value to determine an amount to remove from the outgoing signal with continuous, real-time updating of the threshold value.

Abstract: A method, a system and a computer program product for fast detection of the onset of a near-end signal is provided. An Acoustic Echo Canceller (AEC) attenuates an acoustic echo present in a tele-communication network. The AEC includes an adaptive filter that estimates the acoustic echo and generates an error signal. The error signal is the difference between the acoustic echo and the estimate of acoustic echo plus a near-end signal, if present. The method comprises computing an onset indicator parameter from the error signal and the estimate of acoustic echo. Several other parameters are subsequently calculated by using the onset indicator parameter, the error signal and the estimate of acoustic echo.

Abstract: An apparatus generally having a first circuit and a second circuit is disclosed. The first circuit may be configured to synthesize a first vector by filtering a second vector based on a third vector. The second circuit may be configured to (i) generate a gain corresponding to a fourth vector, (ii) compare the gain to a plurality of thresholds and (iii) update the third vector as a function of the gain where the compare determines that the gain is not between the thresholds. The fourth vector may be received from a network as an echo of the second vector.

Abstract: A digital subscriber line management system is provided. It comprises a device for calculating a set of values specifying a virtual noise mask. The device includes a receiver for receiving or otherwise obtaining a set of measurements of noise levels experienced at a digital subscriber line transceiver at each of a plurality of frequencies at different times, and a processor for calculating, in respect or each frequency, a combined value based on a plurality of measurements taken at different times, and for generating the set of values specifying a virtual noise mask in dependence upon the combined values.

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: An apparatus generally having a first circuit and a second circuit is disclosed. The first circuit may be configured to generate a first sample by filtering an input vector based on (a) a filter vector and (b) a stochastic vector. Each of a plurality of components in the stochastic vector generally has a respective random value. The first circuit may also be configured to generate a second sample as a difference between a third sample and the first sample. The third sample may be received from a network as an echo. The second circuit may be configured to update a subset of a plurality of taps of the filtering where a corresponding one of the components of the stochastic vector has a first value of the random values.

Abstract: A system that incorporates teachings of the present disclosure may include, for example, a controller to: determine crosstalk coupling characteristics between a plurality of lines of a digital subscriber line system connected to a plurality of modems based on a transition between a full power mode and one or more other modes, and provide the determined crosstalk coupling characteristics to one or more of the plurality of modems for performance of at least one of pre-coding a transmitted signal and processing a received signal along a line of the plurality of lines, where the pre-coding and processing is performed based at least in part on the determined crosstalk coupling characteristics, and where the pre-coding of the transmitted signal and the processing of the received signal reduce effects of fluctuating crosstalk. Other embodiments are disclosed.

Abstract: A reduced-memory vectored DSL system includes methods and apparatus for reducing the bandwidth and memory storage demands on a vectored DSL system in which FEXT data is transmitted and stored. When test signal data, such as training and/or tracking data, is sent to determine FEXT characteristics of the DSL system, error signals are available for all or substantially all of the upstream and/or downstream frequency band DSL tones used in the system. Dividing a frequency band into sub-bands, only a subset of tones in each sub-band is used for deriving FEXT data. For tones in the sub-band subsets, full-precision FEXT data values can be derived. For other tones, approximations of the FEXT data can be derived. Memory is reduced in both the transmission of such FEXT data (between upstream and downstream ends of the DSL system) as well as within an upstream-end device such as a DSLAM that performs vectoring using a separate or internal vectoring processing apparatus.

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: The invention provides a method for determining a set of filter coefficients for an acoustic echo compensator in a beamformer arrangement. The acoustic echo compensator compensates for echoes within the beamformed signal. A plurality of sets of filter coefficients for the acoustic echo compensator is provided. Each set of filter coefficients corresponds to one of a predetermined number of steering directions of the beamformer arrangement. The predetermined number of steering directions is equal to or greater than the number of microphones in the microphone array. For a current steering direction, a current set of filter coefficients for the acoustic echo compensator is determined based on the provided sets of filter coefficients.

Abstract: A technique for detecting in-band signaling tones in a communication system includes performing a first adaptation of an adaptive filter of an echo canceller in response to detection of a far-end harmonic signal. In this case, the adaptive filter provides an echo estimation signal. The technique also includes subtracting the echo estimation signal from a near-end signal that includes one or more in-band signaling tones to provide an error signal. The technique further includes detecting, using a tone detector, the one or more in-band signaling tones in the error signal.

Abstract: Methods, apparatuses, and systems are presented for performing adaptive equalization involving receiving a signal originating from a channel associated with inter-symbol interference, filtering the signal using a filter having a plurality of adjustable tap weights to produce a filtered signal, and adaptively updating each of the plurality of adjustable tap weights to a new value to reduce effects of inter-symbol interference, wherein each of the plurality of adjustable tap weights is adaptively updated to take into account a constraint relating to a measure of error in the filtered signal and a constraint relating to group delay associated with the filter. Each of the plurality of adjustable tap weights may be adaptively updated to drive group delay associated with the filter toward a target group delay.

Abstract: A telecommunication system including a speakerphone provides a coupled signal path, including a microphone, configured to sense an incoming audio signal and an echo signal and generate a coupled signal. The echo signal includes non-linear distortion generated by a speaker of the speakerphone. An echo signal path, including an amplifier, is configured to sense the echo signal and generate an echo reference signal. The echo reference signal includes the non-linear distortion. An acoustic echo canceller is configured to receive the coupled signal from the coupled signal path, to receive the echo reference signal from the echo signal path, and to cancel out the non-linear distortion included in the coupled signal based on the non-linear distortion included in the echo reference signal.

Abstract: Method, user device and computer program product for suppressing echo. An audio signal is output from a speaker. A microphone receives an audio signal, wherein the received audio signal includes an echo resulting from the outputted audio signal. A Finite Impulse Response filter estimate ?(n) is dynamically adapted in the time domain based on the outputted audio signal and the received audio signal to model an echo path h(n) related to the echo in the received audio signal. The filter estimate ?(n) is used in an estimate of the echo power in the received audio signal, and the estimated echo power is used to apply echo suppression to the received audio signal, thereby suppressing the echo in the received audio signal.

Abstract: A sound processing apparatus includes: a sound output unit which outputs sound based on a first sound signal as first sound processing when execution of the first sound processing is instructed; a sound collecting unit which generates an echo signal by collecting surrounding sound; a filtering unit which performs filtering process with an adaptive digital filter on the first sound signal and generates a pseudo echo signal; a subtracting unit which removes a sound component, which is output from the sound output unit and reaches the sound collecting unit, from the echo signal by subtracting the pseudo echo signal from the echo signal to generate a residual signal; an updating unit which updates the adaptive digital filter based on the residual signal and the first sound signal; and an update resolution control unit which controls the resolution at which the adaptive digital filter is updated by the updating unit.

Abstract: An exemplary embodiment of the present invention is directed toward a method and system for cancelling line echo in the presence of a known secondary audio signal. Filter adaptation is enabled in the presence of a known secondary audio source such as the sound of a computer game, a music signal or other secondary audio sources that would otherwise prevent echo cancellation due to an apparent double talk condition. It is emphasized that this abstract is provided to comply with the rules requiring an abstract which will allow a searcher or other reader to quickly ascertain the subject matter of the technical disclosure. It is submitted with the understanding that it will not be used to interpret or limit the scope or the meaning of the claims.

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: A method for acoustic echo cancellation in a communication device is provided that includes receiving a first near-end audio signal in the communication device, wherein the first near-end audio signal comprises acoustic echo of a far-end audio signal reproduced by the communication device, and performing echo cancellation on the first near-end audio signal to generate a second near-end audio signal with at least some of the acoustic echo removed, wherein the echo cancellation is performed responsive to presence or absence of double-talk (DT), and wherein a zero-crossing rate (ZCR) is used to detect the presence or absence of DT.

Abstract: The proposed echo canceller comprises:—an adaptive filter for receiving a signal affected by an echo, and for supplying a filtered signal that is an estimate of the echo;—a subtractor for subtracting this estimate from the received signal and supplying a residual signal;—means for detecting (7, 8) a ring back tone in said residual signal;—means for blocking (9) the received signal and replacing it by a locally generated ring back tone if a ring back tone is detected in the received signal;—a timer to determine a time period;—and means (11-17) for, during said time period, replacing the residual signal by some synthetic comfort noise when there is no ring back tone.

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 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 is is converted to the analog from 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. A circuit for acoustic echo cancellation in VoIP terminal contains the digital adaptive filter with the control block situated between the far-end speech signal path and the near-end speech signal path, and the double-talk detector (11) that comprises the signature generator (14) connected by the signature encoder (15).

Abstract: Traditionally, echo cancellation has employed linear adaptive filters to cancel echoes in a two way communication system. The rate of adaptation is often dynamic and varies over time. Disclosed are novel rates of adaptation that perform well in the presence of background noise, during double talk and with echo path changes. Additionally, the echo or residual echo can further be suppressed with non-linear processing performed using joint frequency-time domain processing.