Abstract: A method of acoustic echo suppression using a recurrent neural network, performed by at least one processor, is provided. The method includes receiving a microphone signal and a far-end reference signal, estimating an echo suppressed signal and an echo signal based on the microphone signal and the far-end reference signal, estimating enhancement filters for the microphone signal based on the echo suppressed signal and the echo signal, generating an enhanced signal based on the enhancement filters, and adjusting the enhanced signal using automatic gain control (AGC) and outputting the adjusted signal.

Abstract: A noise reduction apparatus includes: an input device to which an electric signal is input from an external device; a signal processing circuit configured to perform predetermined signal processing on the electric signal input to the input device and output a generated signal to an external device; a switch configured to change over to either one of a connected state in which the input device and the signal processing circuit are electrically connected to each other and a disconnected state in which the input device and the signal processing circuit are electrically disconnected to each other; and a noise processing circuit configured to subtract a noise signal output from the signal processing circuit when the switch is in the disconnected state from the signal output from the signal processing circuit when the switch is in the connected state, and output a subtraction result.

Abstract: A telephone system is provided for transmitting telephone signals between first and second mobile stations. The system includes a first internet protocol interface configured to receive an incoming cell phone signal generated by the first mobile station and to transmit the phone signal to the internet. A second internet protocol interface is configured to receive the phone signal sent through the internet by the first internet protocol interface and to transmit the phone signal to the second mobile station, such that users of the first and second mobile stations can engage in a conversation where the phone signals are communicated over substantial distances through the internet.

Abstract: An audio processing system for a listening device includes an input device, a voice activity detector and a ratio-based attenuator. The input device is configured to receive a first audio signal emanating from an environment and including a signal of interest. The voice activity detector is configured to generate a control signal in response to the first audio signal. The ratio-based attenuator is configured to receive the control signal and determine whether the signal level of the first audio signal exceeds the signal level of an audio signal received from an audio playback device by at least a target difference. If so, then the audio level of the playback audio signal is maintained. Otherwise, the audio level of the playback audio signal is adjusted, where, at the adjusted value, the first signal level exceeds the playback signal level by at least the target difference.

Abstract: A voice processing method and device, the method comprising: detecting a current voice application scenario in a network (S1); determining the voice quality requirement and the network requirement of the current voice application scenario (S2); based on the voice quality requirement and the network requirement, configuring voice processing parameters corresponding to the voice application scenario (S3); and according to the voice processing parameters, conducting voice processing on the voice signals collected in the voice application scenario (S4).

Abstract: A method for multi-channel echo cancellation and noise suppression is described. One of multiple echo estimates is selected for non-linear echo cancellation. Echo notch masking is performed on a noise-suppressed signal based on an echo direction of arrival (DOA) to produce an echo-suppressed signal. Non-linear echo cancellation is performed on the echo-suppressed signal based, at least in part, on the selected echo estimate.

Abstract: A method for echo cancellation in multichannel audio signals includes receiving a plurality of time-domain signals, including multichannel audio signals and at least one reference signal, transforming the time-domain signals to K under-sampled complex-valued subband signals using an analysis filter bank, and performing, for each of the K under-sampled complex-value subband signals, linear echo cancellation of the reference signal from each channel using an acoustic echo canceller. A probability of acoustic echo dominance is produced using a single-double talk estimator, and a semi-blind multichannel source separation is performed based on the probability and independent component analysis (“ICA”) to decompose the audio signals into a near-end source signal and a residual echoes using subband semi-blind source separation.

Abstract: An audio processing system for a listening device includes an input device, a voice activity detector and a ratio-based attenuator. The input device is configured to receive a first audio signal emanating from an environment and including a signal of interest. The voice activity detector is configured to generate a control signal in response to the first audio signal. The ratio-based attenuator is configured to receive the control signal and determine whether the signal level of the first audio signal exceeds the signal level of an audio signal received from an audio playback device by at least a target difference. If so, then the audio level of the playback audio signal is maintained. Otherwise, the audio level of the playback audio signal is adjusted, where, at the adjusted value, the first signal level exceeds the playback signal level by at least the target difference.

Abstract: An apparatus includes an input circuit, a storage device, a retrieve circuit, a decrypt circuit, and an output circuit. The input circuit is configured to receive packets of encrypted encoded video data from a first network. The storage device is preauthorized by the first network to store the encrypted encoded video data. The retrieve circuit is configured to retrieve the encrypted encoded video data from the storage device for decryption. The decrypt circuit is configured to decrypt the encrypted encoded video data without further authorization. The output circuit is configured to transmit a signal representing the encrypted encoded video data while the retrieve circuit retrieves the encrypted encoded video data from the storage device.

Abstract: Methods and devices for enabling and disabling updating an adaptive filter for echo cancellation of a communication device are disclosed. An echo canceller may include an adaptive filter that may be enabled to update responsive to a determination that a first condition is satisfied that there is no double talk detected on the communication device, and a determination that a second condition is satisfied based, at least in part, on the performance characteristic. The adaptive filter may be disabled from updating responsive to at least one of the first condition and the second condition not being satisfied. Disabling the adaptive filter from updating may be responsive to detecting a performance characteristic of the adaptive filter to fail to meet a condition indicating that the adaptive filter is diverging. The condition to be met may be independent of whether double-talk is determined to be present on the communication device.

Abstract: An earpiece (100) and acoustic management module for in-ear canal suppression control are provided. A method for suppressing signals for a conference call, a vehicle, and a general communication event is also provided. The earpiece can include an Ambient Sound Microphone (111) to capture ambient sound, an Ear Canal Receiver (125) to deliver audio content to an ear canal, an Ear Canal Microphone (123) configured to capture internal sound, and a processor (121). The processor can generate a voice activity level (622) and suppress an echo, spoken voice, and media content in the electronic internal signal, and mix an electronic ambient signal with an electronic internal signal in a ratio dependent on the voice activity level and a background noise level to produce a mixed signal (323) that is delivered to the ear canal (131).

Abstract: In order to reduce interference between cells through hopping and use frequencies in a good propagation situation, a scheduler section carries out scheduling for determining to which user data should be sent using CQI from each communication terminal apparatus, selects a user signal to be sent in the next frame and determines in which subcarrier block the data should be sent. An MCS decision section selects a modulation scheme and coding method from the CQI of the selected user signal. A subcarrier block selection section selects a subcarrier block instructed by the scheduler section 102 for each user signal. For the respective subcarrier blocks, FH sequence selection sections select hopping patterns. A subcarrier mapping section maps the user signal and control data to subcarriers according to the selected hopping pattern.

Abstract: A method of reducing noise in an acoustic system, the method comprising at a first user terminal: receiving an audio signal from at least one further user terminal over a network; executing a communication client on a processing unit, the client configured so as when executed to: supply the audio signal to an audio signal processing module of the first user terminal, wherein the processing module processes the audio signal, whereby a level of gain is applied to the audio signal, and outputs a processed audio signal to a speaker; estimate a noise level of the audio signal and the processed audio signal and estimate the gain applied by the processing module taking into account both the noise level estimates; selectively apply a system gain reduction step to at least one of the audio signal and an audio signal received via a microphone, based on at least the estimated gain.

Abstract: A full duplex audio communication terminal (100) comprises a noise attenuation module (4) arranged on a transmission path of the terminal. The noise attenuation module comprises two activation thresholds, which are respectively intended to be adjusted above a noise level and above an echo level of transmission signals (TC) produced by the terminal. The first or the second activation threshold is selected as a function of the detection of signals (R) received by said terminal. An improvement to the suppression in the transmission signals of echoes from received signals is thus obtained which is compatible with the principle of full duplex communication. Preferably, the noise attenuation module processes transmission signals which are produced by an echo compensator (2). Residual echo still present in the transmission signals already processed by the echo compensator is thus further reduced.

Abstract: A device, system and method for enabling the measuring of audiences to media consumed through the use of digital media terminals. An example disclosed herein includes a loop-back device that allows the production/extraction of content identification information from media items rendered by a digital media terminal that can be seamlessly and homogenously implemented in a wide variety of media distribution platforms and devices, including Internet and mobile networks.

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 talk deciding system includes a receiving portion which receives far-end side decision information indicating a talking state on a far-end side, an inputting portion which inputs an emitted signal from the far-end side, a speaker which emits the emitted signal being input into the inputting portion, a microphone which picks up a voice on a near-end side and outputs a picked-up signal, and a deciding portion which preliminarily decides a talking state on the near-end side based on the picked-up signal being output from the microphone and the emitted signal being input into the speaker, and generates near-end side decision information indicating a talking state on the near-end side. The deciding portion decides the talking state on the near-end side based on the far-end side decision information being received at the receiving portion and the near-end side decision information.

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: 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: An input signal is processed through noise suppression (NS) and echo control (EC) via a multipath model that reduces noise pumping effects while maintaining EC performance. A copy of a “noisy” input signal is sent to an EC component before the noisy signal is sent to a NS component, which processes the signal first, when there is a consistent noise level for estimation. The copy of the pre-processing noisy signal is sent to the EC component along with a “clean” or “noise-suppressed” signal output from the NS component. The EC component analyzes the noisy signal as if the EC was the first component in the signal chain to determine what actions to take. The EC component then applies these actions to the clean signal received from the NS component.

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: A method for identifying signals of the same origin is described. The signals are identified based upon a method of system analysis, wherein an accumulated value of an attenuation weighted by power of an incoming channel, the so-called correspondence coefficient, is used as the criterion whether two signals are of the same origin.

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

Abstract: An echo reduction method stores a received audio information stream. A sound detection flag is activated following detection of locally generated sound. Output based on the received audio information stream is muted in response to the activating the sound detection flag. Rendering status of the received audio information stream is saved, in response to the activating the sound detection flag, to reduce loss of audio information. At least a portion of the stored received audio information stream is rendered following inactivation of the sound detection flag.

Abstract: In one embodiment, the present invention is a method for detecting an echo path change (EPC) in a telecommunications network. The method detects whether the effectiveness of echo cancellation of an echo canceller has decreased relatively significantly. Once a relatively significant decrease is detected, the method determines whether the decrease was an EPC or an inadvertent detection of double talk (DT). In particular, the method considers whether echo is effectively cancelled over a hangover period. Further, echo return loss (ERL) estimates are generated over the hangover period and compared to a lowest-possible ERL for the network. If both (1) echo cancellation is ineffective and (2) a sufficient number of ERL estimates are not below the worst-case ERL, then an EPC decision is made. If either (1) echo cancellation is effective or (2) a sufficient number of ERL estimates are below the worst-case ERL, then a DT decision is made.

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: An embodiment of an acoustic echo cancellation system is disclosed. The system comprises an echo cancellation unit, a second filter and a subtraction unit. The echo cancellation unit comprises a first attenuator, a first filter and a first subtractor. The first attenuator has a first down-scaling factor for attenuating a first signal. The first filter generates a first echo signal estimate based on the attenuated first signal. The first subtractor generates a third signal by subtracting the first echo signal estimate from a second signal. The second filter generates a second echo signal estimate based on the first signal. The subtraction unit subtracts the second echo signal estimate from the third signal.

Abstract: In one embodiment, a high-level compensation (HLC) module receives samples of an input signal and determines whether a magnitude of each sample, represented in a linear domain, is relatively low or relatively high by comparing the magnitude to a threshold. If a sample is less than or equal to the threshold, then it is considered to have a relatively low magnitude and the sample is not attenuated. If a sample is greater than the threshold, then it is considered to have a relatively high magnitude and the HLC module attenuates the sample according to a “soft” non-linear function. The “soft” non-linear function is characterized by at least two of the following characteristics: the non-linear function (i) increases monotonically, (ii) forms a convex upwards curve, (iii) has a first derivative at the threshold equal to one, and (iv) has a first derivative at a maximum possible magnitude value equal to zero.

Abstract: A state deciding unit 2 makes a decision of double talk when both received signal and transmission signal are in a speech burst, and when the amount of reduction in an echo removal level per prescribed time period exceeds a predetermined threshold, the echo removal level being represented as a level difference between the transmission signal and a residual signal passing through the echo elimination by an adaptive filter 1, and when the amount of changes of the signal level of the received signal is less than a prescribed reference value.

Abstract: Systems and methods for blind subband acoustic echo cancellation postfiltering in a communication device are provided. In exemplary embodiments, an acoustic signal is received via a microphone of the communication device. Acoustic echo cancellation (AEC) is applied to the acoustic signal to obtain an AEC masked signal. Because residual echo may still exist in the AEC masked signal, blind subband AEC postfiltering on the AEC masked signal may be performed to obtain an echo-free acoustic signal. The echo-free signal may then be output.

Abstract: A communication system includes transmit and receive paths, a balancing filter, a noise generator, a detector, and an interface. The transmit path is operable to generate data for communication on a communication network. The receive path is operable to receive data from the communication network. The balancing filter is coupled between the transmit path and the receive path. The noise generator is operable to inject a noise signal on the receive path. The detector is operable to measure reflected power in the transmit path associated with the noise signal. The interface is operable to receive a plurality of sets of coefficients for configuring the balancing filter, wherein the detector is operable to measure the reflected power for each of the sets of coefficients.

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

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

Abstract: An apparatus and method for controlling acoustic echo cancellation in a mobile communication terminal are capable of improving phenomena occurring during a call, such as an echo phenomenon, muting phenomenon and fluctuation of transmitting signals, by performing real-time monitoring of the signal of a speaker output stage and effectively controlling an AEC function according to the characteristics of the terminal. The level of a receiving signal which is outputted to a speaker is detected. A transmitting signal, which is outputted from a microphone, is converted into a digital voice signal. A certain attenuation value is adaptively determined depending on the detected level of the receiving signal, and attenuation on transmitting signal is controlled according to the determined attenuation value.

Abstract: A method for conversation like rendering of a stored audio information stream determines a first location in the stored audio information stream. The first location represents a point in time when the sound detection flag became active. The method next moves from the first location to a second location in the stored audio information stream. The second location is selected based upon a criterion to make playback of the stored audio information stream appear like actual conversation. Finally, the stored audio information stream is rendered starting with audio information stored at the second location.

Abstract: A system for limiting a received audio signal in a communication system is provided. The receive audio signal is limited prior to being played over a loudspeaker to insure that the loudspeaker output will not be clipped when picked up by a nearby microphone associated with the communication system. By preventing clipping of the loudspeaker output at the microphone, the transfer function of the loudspeaker-enclosure-microphone system remains linear, facilitating accurate echo cancellation in the communication system.

Abstract: This invention provides a technique to reduce echoes not adequately attenuated by echo cancellers during initial periods of voice communication. Signals transmitted across communication networks are often reflected back to the caller resulting in echo. Echo cancellers are employed in the communication system to cancel this effect in order to maintain a high quality transmission. However, echo cancellers require time to detect, adapt, and effectively remove echo, often resulting in echo during the initial moments of the call and thereby degrading the quality of service. By installing an attenuation device into the communication system, all signals that pass through it are reduced to a preset value for a set period of time. This reduces any echo below a detectable threshold. After the period of time expires, the attenuation device allows the signal to pass unaffected, by which time the echo cancellers have adaptively adjusted to any echoes in the system.

Abstract: A received power calculating section and a sending power calculating section calculate a received power value Rcv_Power(t) and a sending power value Snd_Power(t) by respectively adding predetermined gain values when the received power calculating section and the sending power calculating section calculate the received power value Rcv_power(t) and the sending power value Snd_Power(t). A received detection value control section and a sending detection value control section control the calculated receiving detection value and the sending detection value and set the received detection value and the sending detection value so as not to become less than a predetermined lower limit value, respectively.

Abstract: A method (200) of cancelling echo in a duplex communication device (100). The method can include detecting a level of noise present on an uplink signal path (104), generating a noise classifier (194) based on the detected level of noise, detecting whether uplink audio is present on the uplink signal path (104) and detecting whether downlink audio is present on a downlink signal path (102). The method further can include generating a double talk flag (136) based at least on the noise classifier, whether uplink audio is present on the uplink signal path, and whether downlink audio is present on the downlink signal path. In addition, the double talk flag, the noise classifier and an uplink signal can be processed to generate an output signal (120) having reduced echo.

Abstract: A method and corresponding apparatus for managing voice quality by optimizing the amount of attenuation placed in the echo return path based on the conditions observed during each telephone call. At the beginning of a telephone call, a pre-configured initial amount of attenuation is placed in the echo path of a communications connection that includes an echo canceller. After the call has begun, ERL is measured at a point in the communications connection that includes the attenuation. If the measured ERL is higher than necessary for the echo canceller to accurately perform double-talk detection, the attenuation in the echo path is slowly reduced from the initial amount. As the attenuation is reduced, the measured ERL reduces accordingly. When the measured ERL reduces to a minimum value required to perform double-talk detection, the reduction in attenuation is stopped.

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

Abstract: A method of determining residue-echo threshold for the trans-hybrid echo path of a loop start (LS) trunk line, comprising the steps of estimating the impulse response of the trans-hybrid echo path, passing a speech-like signal through the LS trunk circuit and collecting the echo signal. The collected echo signal is compared to the convolution of the speech-like signal with the estimated impulse response. The difference between the convolution result and the collected echo signal is the residue echo, which represents the effect of nonlinearity. With different levels of excitation signal, a curve of residue echo power level against input signal level is obtained, and the distortion characteristics are derived from this curve.

Abstract: This device for processing audio signals for telecommunication equipment comprises automatic gain control means (18) acting on the level of the received audio signal (Sr) by amplifying and attenuating the signal as a function of the level of the received signal and echo cancelling means (22). The automatic gain control means (18) comprise amplification means (19-b) and attenuation means (19-a) for separately amplifying and attenuating the received signal downstream and upstream of the echo cancelling means (22), when considering the propagation direction of the received signal.

Abstract: A communication system includes transmit and receive paths, a balancing filter, a noise generator, a detector, and an interface. The transmit path is operable to generate data for communication on a communication network. The receive path is operable to receive data from the communication network. The balancing filter is coupled between the transmit path and the receive path. The noise generator is operable to inject a noise signal on the receive path. The detector is operable to measure reflected power in the transmit path associated with the noise signal. The interface is operable to receive a plurality of sets of coefficients for configuring the balancing filter, wherein the detector is operable to measure the reflected power for each of the sets of coefficients.

Abstract: A path change is detected by multiplying the energy of a signal on the output of a summation circuit in one channel of a telephone by a constant to produce a product. The product is compared with the energy of a signal on the input of the summation circuit. A path change is indicated when the energy of the product exceeds the energy of a signal on the input of the summation circuit. The comparison is made in successive frames of an audio signal and a path change is indicated when the energy of a signal on the input of the summation circuit is exceeded in two or more successive frames.

Abstract: An echo canceler removes the echo of a receive signal from a transmit signal while also controlling the signal level of the transmit signal. The echo canceler generates signal level data for the transmit signal, updates the signal level data when the transmit signal is active and the receive signal is inactive, and amplifies the transmit signal according to the signal level data. The echo canceler also generates an echo cancellation signal from the receive signal, amplifies the echo cancellation signal according to the signal level data, and subtracts the amplified echo cancellation signal from the amplified transmit signal to obtain a transmit output signal.

Abstract: A method and apparatus for adapting dual filters is disclosed. In one aspect, a method may include transforming a signal, adapting a first adaptive filter based on the transformed signal, estimating a delay of an impulse response based on the adaptation of the first filter, delaying a signal based on the estimated delay, and adapting a second adaptive filter based on the delayed signal. In one aspect, an echo or other unwanted signal may be reduced or cancelled based on the adaptation of the second filter.

Abstract: An echo canceller used in a vocal information exchange system operates in such a circumstance that a sound wave released by a speaker is picked up by a microphone unintentionally and works to suppress an echo signal included in the audio signal produced by the microphone by using a pseudo echo signal which is produced based on the incoming audio signal to be fed to the speaker and produce a subtracted audio signal which becomes the outgoing audio signal, with the echo signal being suppressed based on the accurately-calculated degree of echo suppression achieved by use of the pseudo echo signal.

Abstract: The invention relates to an echo canceling device and method for an arrangement for transmitting audio signals, in particular uttered speech with an echo filter for canceling echo. The echo filter is arranged between an input channel (1) and an output channel (4). The input channel (1) leads a first electrical input signal to a converter (2) for converting the electrical signal into a first audio signal. The output channel (4) transmits an electrical signal from a converter (3) which converts a second audio signal back into an electrical signal. In order to prevent non-linearly distorted echo in such an echo canceling device or method, a high-pass filter (8) is arranged in the input channel (1) which high-pass filter (8) has a cut-off frequency that is beyond the cut-off frequency of the high-pass behavior in the converter (2). The high-pass filter (8) is followed by a limiting element (9) which limits the amplitude of the input signal to the input range of the converter (2).

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.