Abstract: A sound processor includes a conversion unit converts a reference sound signal corresponding to a base of sound to be output and an observation sound signal based on each of sound signals output by a plurality of sound receiving units into frequency components, an echo suppression unit estimates echo derived from sound based on a converted reference sound signal and suppressing the estimated echo in a converted observation sound signal, a noise suppression unit estimates noise based on an arrival direction of sound and suppressing the estimated noise in the converted observation sound signal and an integrating process unit suppresses, with respect to each frequency component, echo and noise in the converted sound signal based on a observation sound signal obtained after echo suppression and a observation sound signal obtained after noise suppression.

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: Converter 100 uses either the output signal of a sound pickup device or the signal obtained by subtracting the output signal of an echo canceller from the output signal of the sound pickup device as a first signal, uses an estimated crosstalk value indicative of an estimated value of the amount of crosstalk of an echo leaking into the first signal to correct the first signal, and limits the corrected first signal not to be smaller than estimated near-end noise.

Abstract: A regression-based residual echo suppression (RES) system and process for suppressing the portion of the microphone signal corresponding to a playback of a speaker audio signal that was not suppressed by an acoustic echo canceller (AEC). In general, a prescribed regression technique is used between a prescribed spectral attribute of multiple past and present, fixed-length, periods (e.g., frames) of the speaker signal and the same spectral attribute of a current period (e.g., frame) of the echo residual in the output of the AEC. This automatically takes into consideration the correlation between the time periods of the speaker signal. The parameters of the regression can be easily tracked using adaptive methods. Multiple applications of RES can be used to produce better results and this system and process can be applied to stereo-RES as well.

Abstract: Implementations related to echo cancellation are depicted and described herein.

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

Abstract: A multichannel acoustic echo reduction system is described herein. The system includes an acoustic echo canceller (AEC) component having a fixed filter for each respective combination of loudspeaker and microphone signals and having an adaptive filter for each microphone signal. For each microphone signal, the AEC component modifies the microphone signal to reduce contributions from the outputs of the loudspeakers based at least in part on the respective adaptive filter associated with the microphone signal and the set of fixed filters associated with the respective microphone signal.

Abstract: This invention proposed an Echo Suppressor which can efficiently suppress both echoes and background noise without introducing “choppiness”. The Echo Suppressor System includes said two adaptive gains Gr(RSR) and Gn(NSR), said one adaptive zeros-filter A1(z) and said one adaptive poles-filter A2(z); wherein, thr gain Gr(RSR) is controlled by RSR (Residual echo level to Signal level Ratio); the gain Gn(NSR) is controlled by NSR (Noise signal level to current Signal (Tx) level Ratio); the filter A1(z) is converted from LSF1 obtained from the first modification of LSFTx (Line Spectral Frequencies of Tx signal); the filter A2(z) is converted from LSF2 obtained from the second modification of LSFTx.

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

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

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

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

Abstract: In a closed loop acoustic system with forward and return paths, gain is controlled with different instantiations of a common gain control engine on each path. The gain control engine computes a first gain based on near-end signal levels, computes a second gain based on the output of the gain control engine, computes a third gain based on a howling condition, and computes a final gain as the product of the first, second and third gains.

Abstract: An electronic device comprises a microphone, a transceiver circuit, a loudspeaker, a sense element and a processing circuit. The microphone is configured to receive a first audio signal. The transceiver circuit is configured to communicate the first audio signal to a remote device and to receive a second audio signal from the remote device. The loudspeaker is configured to provide an audible signal based on the second audio signal. The sense element is configured to sense the audible signal provided by the loudspeaker. The sense element may be positioned at a distance from the loudspeaker different than a distance between the microphone and the loudspeaker. The processing circuit is configured to process at least one of the first audio signal and the second audio signal based on a sensed signal from the sense element.

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 system and method for enhancing communications through a phone. A voice communication is received from a user of the phone. A secondary signal is received from an environment in proximity to the phone. The secondary signal is processed to determine an inverse signal in response to receiving the secondary signal. The inverse signal is combined with the voice communication and the secondary signal to destructively interfere with the secondary signal for allowing a receiving party to more effectively communicate with the user.

Abstract: In a data processing system, a method includes receiving samples of a send path input signal (Sin) and samples of a receive path input signal (Rin); in response to receiving each sample of the send path input signal, providing a double-talk indicator to indicate whether or not double-talk has been detected; using the double-talk indicator to determine a double-talk density within a predetermined number of samples of the send path input signal or the receive path input signal; and using the double-talk density to provide a heavy double-talk indicator which indicates whether or not heavy double-talk has been detected. In response to the heavy double-talk indicator indicating that heavy double-talk has been detected, a heavy double-talk mode may be entered.

Abstract: A linear full duplex system and method for acoustic echo cancellation is disclosed. In one embodiment, a method includes calculating a residual echo after subtraction of an echo estimate from a near end signal associated with a communication, refining a far end and a residual signal associated with the communication, updating, based on the far end signal, the echo estimate of an echo associated with the communication, subtracting the echo from the near end signal based on the echo estimate to cancel the echo associated with the communication, updating, based on the refined far end and refined residual signal, the adaptive filter module used for echo estimation, and detecting a steady state and, during the steady state, dynamically detecting internal substates and switching between the internal substates and detecting and managing double talk associated with the communication.

Abstract: A method and system for eliminating echo in a speaker-microphone communication system are provided.

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 communication device includes memory, an input interface, a processing module, and a transmitter. The processing module receives a digital signal from the input interface, wherein the digital signal includes a desired digital signal component and an undesired digital signal component. The processing module identifies one of a plurality of codebooks based on the undesired digital signal component. The processing module then identifies a codebook entry from the one of the plurality of codebooks based on the desired digital signal component to produce a selected codebook entry. The processing module then generates a coded signal based on the selected codebook entry, wherein the coded signal includes a substantially unattenuated representation of the desired digital signal component and an attenuated representation of the undesired digital signal component. The transmitter converts the coded signal into an outbound signal in accordance with a signaling protocol and transmits it.

Abstract: Harmonic distortion residual echo suppression (HDRES) technique embodiments are presented which act to suppress the residual echo remaining after a near-end microphone signal has undergone AEC, including harmonic distortion in the signal that was caused by the speaker audio signal playback. In general, an AEC module is employed which suppresses some parts of the speaker audio signal found in a near-end microphone signal and generates an AEC output signal. A HDRES module then inputs the AEC output signal and the speaker audio signal, and suppresses at least a portion of a residual part of the speaker audio signal that was left unsuppressed by the AEC module. This includes at least a portion of the harmonic distortion exhibited in the AEC output signal.

Abstract: Acoustic echo cancellation methods are provided, in which first and second suppression gains of first and second sub-bands of an Nth acoustic frame are generated, and the second suppression gain of the second sub-band of the Nth acoustic frame is adjusted according to the first suppression gain of the first sub-band of the Nth acoustic frame.

Abstract: A voice communication device with an integrated framework structure for echo cancellation and noise reduction is disclosed. A microphone receives a local input signal while a speaker is outputting a local output signal. The local input signal and output signal are all decomposed into a plurality of subband signals by filter banks for conducting individual processing of echo cancellation and noise reduction per subband. The subband echo canceller is followed by a DFT unit to split the cancellation result into a plurality of narrow frequency bins whereby the noise reduction is performed. The noise reduction results are recombined by an IDFT unit for residual echo removal in a subband non-linear processor. The final output is obtained from a synthesis filter bank that synthesizes the subband signals after echo cancellation and noise reduction into a full-band signal.

Abstract: An acoustic echo cancellation device for canceling an echo in a microphone signal in response to first and second input signals includes a first combination unit for combining the first and second input signals into an aggregate input signal. The device further includes an adaptive filter unit for filtering the aggregate input signal so as to produce an aggregate echo cancellation signal. A second combination unit combines the aggregate echo cancellation signal with the microphone signal so as to produce a residual signal, and an additional filter unit filters either the first or second input signal so as to produce a first or a second partial echo cancellation signal. A post-processor unit suppresses remaining echo components in the residual signal. The post-processor unit uses at least one partial echo cancellation signal to suppress echo components corresponding with the first input signal to a greater extent than echo components corresponding with the second input signal.

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: An apparatus and method for interference cancellation is provided to cancel the interference such as echo and cross-talk received by a receiver of a communication system. The apparatus includes a digital cancellation signal generator, a first canceller, and a second canceller. The digital cancellation signal generator can generate a digital cancellation signal, which includes a first and a second portion and represents an interference signal within a received signal. The first canceller can perform an analog cancellation on the received signal to output a partially-interference-canceled received signal according to the first portion of the digital cancellation signal. The second canceller can perform a digital cancellation on the partially-interference-canceled received signal according to the second portion of the digital cancellation signal.

Abstract: An acoustic echo cancellation device (1) for canceling an echo in a microphone signal (z) in response to a far-end signal (x) comprises: an adaptive filter unit (10) arranged for filtering the far-end signal (x) so as to produce an echo cancellation signal (y), —a combination unit (12) arranged for combining the echo cancellation signal (y) with the microphone signal (z) so as to produce a residual signal (r), and a post-processor unit (13) arranged for substantially removing any remaining echoes from the residual signal. In accordance with the invention, the device further comprises: —a near-end pitch estimation unit (18) arranged for estimating the pitch of any near-end signal (s) contained in the microphone signal (z) and for controlling the post-processor unit (13) in dependence of the estimated pitch such that harmonics of the estimated pitch may be substantially preserved.

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: Described is automatically testing the quality of an audio coupling between juxtaposed first and second digital telephones, e.g., VoIP telephones, such as to quantitatively determine the quality of audio echo cancellers in those digital telephones. An analyzer receives timestamps from a first telephone and second telephone during a calling session, including timestamps for when the second telephone initially provides audio (e.g., speech) to the first telephone, when the first telephone initially detects sound, when the first telephone initially provides audio to the second telephone, and when the second telephone initially detects sound. The analyzer uses the relative timing of the timestamps and the speech recognizer's outcome to determine whether the audio coupling is experiencing interference or echo. When the audio includes speech, a confidence level corresponding to accuracy of speech recognition also may establish the audio coupling's quality.

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

Abstract: In a communication machine room wideband noise suppression system, a sensing unit detects a noise source produced by a fan during the operation thereof and generates a noise input signal and a feedback signal, which are sent to a signal amplifying unit for signal amplification. The amplified signals are then sent to a signal converting unit and converted into digital signals. A multi-channel hybrid controller receives the digital signals and makes corrections and conduct rapid convergence algorithm to derive a reverse digital signal, which is sent to the signal converting unit and converted into a reverse analog signal. The reverse analog signal is sent to the signal amplifying unit for power amplification to generate a control signal for driving a loudspeaker unit to produce interfering acoustic wave, so as to cancel out the noise source and achieve the purpose of eliminating wideband noise.

Abstract: Duplex audio communications over a network use compressed audio data, with linear prediction coefficients (LPCs) and variances by which sample values differ from predictions. A adaptive echo canceller for a transceiver develops finite impulse response filter (FIR) coefficients characterizing an echo path between its local audio output and audio input. The received/decompressed audio data is applied to the FIR coefficients, and the predicted echo is subtracted from the uplink signal. Echo is detected as cross-correlation of the receive signal versus the uplink/send signal over time. In one embodiment, the cross-correlation is determined using a pre-whitened receive signal, obtained by adopting the variance values received over the network by the downlink Codec. Apart from the uplink Codec, no speech analysis filter or process is needed. The technique is apt for GSM, AMR and similar compressed audio communications.

Abstract: A device comprising a circuit for picking up acoustic signals with the microphone; a circuit for playing back audio signals with an amplifier and a loudspeaker; an interface circuit for interfacing with a telephone network or a cell phone; and a circuit for digitally processing audio signals and reducing the acoustic echo that results from interaction of the loudspeaker with the microphone. The circuit for digitally processing audio signals comprises: an echo cancellation stage with an adaptive linear filter suitable for subtracting from the signal picked up by the microphone a reference signal that is derived from the signal received at the output from the interface; a gain control stage for suppressing the residual echo, and a stage for selectively reducing the background noise present in the signal received at the output from the echo suppression stage.

Abstract: The device comprises a microphone for detecting a speech signal from a near speaker, and a loudspeaker for reproducing a speech signal from a remote speaker. The processing for canceling the interfering acoustic echo implements an adaptive linear filtering algorithm. Double talk situations are detected by: evaluating an index representative of the convergence or divergence of the algorithm; assessing a predetermined condition for detecting a double talk situation; and if the condition is satisfied, modifying at least one parameter of the algorithm in response to the detection. The representative index may be the norm of the gradient vector describing the adaptation of the filter from one iteration of the algorithm to the next, the conditions being a comparison between the gradient and a threshold. The parameter that is modified double talk situation may be the adaptation stepsize of the algorithm, and also the gain control of an echo suppression stage.

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: A hybrid circuit for a full-duplex transceiver includes a two-to-four wire ratio converter (50, Z1, Z2) for forwarding a transmitted signal TX? on a transmission line (41) in response to an input signal TX. The ratio converter also produces a signal RXA in response to a received signal RX arriving on the transmission line, wherein signal RXA includes an echo of input signal TX. A passive filter (H1) and a first active filter (H2) each filter input signal TX to generate signals TXA and TXB summed with the RXA signal to form a signal RXB at the input of an amplifier (52, Z3, Z4) producing an output signal RXC. A second active filter (H3) filters input signal TX to generate a signal TXC summed with the RXC signal to produce the output signal RX. The impedance of the passive filter is designed so that when the transmission line has a target impedance, such as the impedance of an ideal twisted pair, the TXA output signal of the passive filter adequately offsets the TX signal echo in RXA.

Abstract: An audio interface adapted to reduce a subscriber voice may receive a subscriber voice and a background noise. The subscriber voice may then be compared to the to the background noise. If the received subscriber voice is louder than the received background noise, the audio interface may output a message to the cellular telephone subscriber indicating the subscriber may reduce his speaking volume. Additionally, an audio interface may process a voice waveform that corresponds to the subscriber voice and a background waveform that corresponds to the background noise to generate a substantially opposite voice waveform and a substantially opposite background waveform respectively. The substantially opposite voice waveform and background waveform may be substantially out of phase from the voice waveform and background waveform respectively and output via one or more output ports of the audio interface.

Abstract: An echo cancellation technique that can process multi-input microphone signals with only a small increase in the overall CPU consumption compared to implementing the algorithm for a single channel microphone signal. Furthermore, the invention provides an architecture that provides for echo cancellation for multiple applications in parallel with only a small increase in CPU consumption compared to a single instance of echo cancellation with a single microphone input and multi-output channel playback.

Abstract: Duplex audio communications over a network use compressed audio data, with linear prediction coefficients (LPCs) and variances by which sample values differ from predictions. A adaptive echo canceller for a transceiver develops finite impulse response filter (FIR) coefficients characterizing an echo path between its local audio output and audio input. The received/decompressed audio data is applied to the FIR coefficients, and the predicted echo is subtracted from the uplink signal. Echo is detected as cross-correlation of the receive signal versus the uplink/send signal over time. In one embodiment, the cross-correlation is determined using a pre-whitened receive signal, obtained by adopting the variance values received over the network by the downlink Codec. Apart from the uplink Codec, no speech analysis filter or process is needed. The technique is apt for GSM, AMR and similar compressed audio communications.

Abstract: An adaptive howling canceller has a plurality of adaptive filters. A delay adds a time delay of an acoustic feedback path to an electric signal fed from an amplifier of a sound-reinforcement system. Each adaptive filter filters the output signal of the delay with a filter coefficient, which is periodically updated at an update interval. The update interval of each adaptive filter is set to decrease successively from a first one to a last one of the adaptive filters. Adders are arranged in correspondence to the adaptive filters in series between a microphone and the amplifier. Each adder subtracts the output signal of the corresponding adaptive filter from an output signal fed from a preceding adder to thereby provide an output signal to a succeeding adder. The output signal from each adder is inputted into the corresponding adaptive filter.

Abstract: An echo canceller (106) can include a first multi-band filter (152) which receives a first input signal (108) and generates a first plurality of sub-band signals (110, 111, 112), and a second multi-band filter (154) which receives a second input signal (122) and generates a second plurality of sub-band signals (156, 157, 158). The echo canceller also can include a plurality of double talk detectors (168, 169, 170) that each generate a double talk flag (182, 183, 184) based on at least a respective one of the first sub-band signals and a respective one of the second sub-band signals.

Abstract: Providing echo canceling apparatus, telephone set using the same and an echo canceling method that do not require transmission/reception sound detection and are resistant to environmental noise. Echo canceling apparatus according to the invention includes a calculator 4 for subtracting a pseudo-echo signal pec generated from a reception sound signal from an echo signal ec that is based on an echo generated when the reception sound that is based on the reception sound signal is reflected on a reflective body such as a wall thereby outputting a residual echo signal rec. The echo canceling apparatus further includes a residual echo detector 12 for estimating a residual echo volume by detecting sign inversion of the echo signal ec and the residual echo signal rec and their amplitude values.

Abstract: A regression-based residual echo suppression (RES) system and process for suppressing the portion of the microphone signal corresponding to a playback of a speaker audio signal that was not suppressed by an acoustic echo canceller (AEC). In general, a prescribed regression technique is used between a prescribed spectral attribute of multiple past and present, fixed-length, periods (e.g., frames) of the speaker signal and the same spectral attribute of a current period (e.g., frame) of the echo residual in the output of the AEC. This automatically takes into consideration the correlation between the time periods of the speaker signal. The parameters of the regression can be easily tracked using adaptive methods. Multiple applications of RES can be used to produce better results and this system and process can be applied to stereo-RES as well.

Abstract: Architecture that mitigates echo in voice communications using echo detection and adaptive management of attenuation by a non-linear processor (NLP). Suppression values provided by the NLP are determined based on echo detection and retained on a case-by-case basis to automatically increase or decrease the attenuation as needed. Feedback is incorporated that where the controls for the NLP attenuation is given by the amount of echo that remains, and this in turn affects the amount of echo that remains.

Abstract: A system and method for estimating a channel spectrum. The method includes: (a) receiving an input signal from a channel, where the input signal includes one or more major echoes and zero or more minor echoes introduced by the channel; (b) identifying the one or more major echoes present in the input signal; (c) identifying the minor echoes from a filtered autocorrelation function of the input signal in response to a determination that there is only one major echo; (d) identifying the minor echoes from a filtered power spectrum of the input signal in response to a determination that there is more than one major echo; (e) computing a channel spectrum estimate from the major echoes and minor echoes; where the channel spectrum estimate is usable to remove at least a portion of the one or more major echoes and one or more minor echoes from the input signal.

Abstract: Methods and apparatus are provided for reducing echo from a received signal. A suppression gain is applied to an output of an echo canceler that has processed the received signal. The suppression gain includes a region of sloping attenuation about a decision point. The echo canceler optionally estimates an echo path and subtracts then estimate from the received signal. The suppression gain includes a non-zero lower bound, gmin, on a maximum attenuation applied by a suppressor, that is based on operating conditions of the echo canceler. The region of sloping attenuation applies a variable amount of attenuation that depends on the size of an output error ?(n) of the echo canceler relative to the received signal, x(n).

Abstract: A system processes signals exchanged between a near end terminal and a far end terminal over a communication path. The system selects the processing characteristics of a near end terminal based on characteristics of the communication path. The communication path may include the near and the far end terminal and their respective codecs, and the networks. The system may select processing characteristics of the near end terminal based on characteristics of the communication path. Selecting the processing characteristics may include selecting a predetermined operation. The system may adapt a selected operation based on a real time analysis of the communication path.

Abstract: In an acoustic echocanceller (6), an estimate of an echo signal is determined by an adaptive filter (10) and is subtracted from the input signal by a subtracter (14). The spectrum estimator (12) determines the frequency spectrum of the estimate of the echo signal, and the filter (16) filters the output signal of the subtracter (14) with a filter having a transfer function dependent on the spectrum determined by the estimator (12). The use of this combination results in a substantial improvement of the suppression of the echo signal.