Abstract: An AEC system can implement different updating techniques for echo cancellation. A Block Maximum Likelihood updating technique updates filter coefficients typically at integer multiples of a block size. A Proportional-Incremental Maximum Likelihood updating technique applies proportional weighting to the updates of filter coefficients. A Multi-source Incremental Maximum Likelihood updating technique uses separate control parameters associated with each of multiple loudspeakers. Combinations of these may be used, and one or more of these can be selected and implemented in real-time.

Abstract: There is provided an echo suppression device that allows adding a natural comfort noise even in an environment in which a change in ambient noise is large. An echo is removed from an input signal to generate an echo-removed signal, a noise signal included in the echo-removed signal is estimated to generate an estimated noise signal, and a noise signal included in the echo-removed signal is suppressed based on the estimated noise signal to generate a suppressed signal. A comfort noise is generated based on the estimated noise signal, and the comfort noise is superimposed on the suppressed signal.

Abstract: An audio device is provided. The audio device includes processing circuitry which is connected to a loudspeaker and a microphone. The processing circuitry is configured to play an echo reference signal from a far end on the loudspeaker, and perform an acoustic echo cancellation (AEC) process using the echo reference signal and an acoustic signal received by the microphone using an AEC adaptive filter. The processing circuitry repeatedly determines a first status of the loudspeaker according to a relation between the played echo reference signal and the received acoustic signal, and transmits a first status signal indicating the first status of the loudspeaker to the far end through a cloud network.

Abstract: An echo cancellation detector for controlling an acoustic echo canceller that is configured to cancel an echo of a far-end signal in a near-end signal in a telephony system, the echo cancellation detector comprising a comparison generator configured to compare the far-end signal with the near-end signal, a decision unit configured to make a determination about a first acoustic echo canceller based on that comparison and a controller configured to control an operation of a second acoustic echo canceller in dependence on the determination.

Abstract: An interference canceling subsystem for a bidirectional communications network includes an input interface configured to receive a first data signal from a first transceiver of the network, an output portion configured to receive a second data signal from a second transceiver of the network, a first signal path connecting the input interface to the output portion, a second signal path connecting the output portion to the input interface, and a first interference canceler disposed between the output portion and the input interface along the second signal path. The first signal path is configured to relay the first data signal from the input interface to the output portion. The interference canceler is configured to (i) relay the second data signal from the output portion to the input interface, and (ii) remove portions of the first data signal from the relayed second data signal prior to reaching the input interface.

Abstract: A method and a device for updating a coefficient vector of a finite impulse response filter are provided. The update method includes: obtaining an updated step-size diagonal matrix for a coefficient vector of the FIR filter; and obtaining an updated coefficient vector of the FIR filter based on the updated step-size diagonal matrix.

Abstract: A system configured to improve audio processing by performing dereverberation and noise reduction during a communication session. The system may apply a two-channel dereverberation algorithm by calculating coherence-to-diffuse ratio (CDR) values and calculating dereverberation (DER) gain values based on the CDR values. While the DER gain values may be calculated at a first stage within the pipeline, the device may apply the DER gain values at a second stage within the pipeline. For example, the device may calculate the DER gain values prior to performing residual echo suppression (RES) processing but may apply the DER gain values after performing RES processing, in order to avoid excessive attenuation of the local speech. In addition to removing reverberation, the DER gain values also remove diffuse noise components, reducing an amount of noise reduction required. Thus, the device may soften noise reduction when the DER gain values are applied.

Abstract: In a sound signal processing apparatus, a communication part receives a far-end sound signal from a smartphone serving as a sound source. A correlation value calculator calculates a correlation value indicative of a correlation between the far-end sound signal and a sending sound signal that includes an echo signal deriving from the far-end sound signal. When the correlation value is equal to or smaller than a predetermined suppression threshold, a determiner determines to suppress the far-end sound signal. When the determiner has determined to suppress the far-end sound signal, a suppressor suppresses the far-end sound signal, and outputs the suppressed far-end sound signal to a sound volume adjuster.

Abstract: An electronic device and a controlling method therefor are provided. The electronic device includes a speaker, a microphone, and an audio processor configured to adjust a size of a signal of a predetermined frequency band in an input audio signal, determine whether to adjust the size of the audio signal wherein the size of the frequency band was adjusted based on the output level of the speaker, and output the audio signal processed based on whether the adjustment was performed through the speaker. The audio processor is further configured to perform acoustic echo cancellation for the sound signal using the input audio signal or the audio signal of which size was adjusted based on whether the adjustment was performed, based on receiving a sound signal including the output audio signal through the microphone.

Abstract: A technique for reducing acoustic feedback in audio communications includes measuring variations in round-trip delay over an audio signal pathway. The technique varies a delay interval of an adjustable-delay element in real time based on the measured variations in round-trip delay, effectively canceling the delay variations. Further techniques are disclosed for detecting and eliminating howling frequencies which arise as a result of acoustic feedback in the audio signal pathway.

Abstract: Enabling and disabling the updating of an adaptive filter for echo cancellation of a communication device is discussed. In some embodiments, an echo canceler in a communication device may include an adaptive filter 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 a 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. Described is disabling the adaptive filter from updating 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 may be independent of whether double-talk is determined to be present on the communication device.

Abstract: A gain control system for dynamically tuning an echo canceller, the echo canceller being configured to estimate an echo of a far-end signal and subtract that echo estimate from a microphone signal to output an echo cancelled signal, the gain control system comprising a monitoring unit configured to estimate an energy associated with an impulse response of an adaptive filter configured to generate the echo estimate from the far-end signal and a gain tuner configured to adjust an attenuation of at least one of the microphone signal and the far-end signal in dependence on the estimated energy.

Abstract: An audio processing device includes a first microphone configured to receive a first signal a second microphone configured to receive a second signal, a noise reduction gain determination circuit configured to determine a noise reduction gain based on the first signal and the second signal, a noise reduction circuit configured to attenuate the first signal based on the determined noise reduction gain, and an output circuit configured to output the attenuated signal.

Abstract: Embodiments of the present invention provide a crosstalk suppression method and apparatus. By obtaining a first electrical signal indicating a data bit stream of a first optical signal output by a laser diode (LD) in an optical transceiver module and a transmission parameter of a photodiode (PD), adjusting the first electrical signal based on the transmission parameter of the PD to obtain a third electrical signal indicating an electrical signal transformed by the PD from the first optical signal transmitted by the LD, and subtracting the third electrical signal from a second electrical signal output by the PD in the optical transceiver module, an electrical signal generated by the optical signal transmitted by the LD and reflected or refracted to the PD is removed from the electrical signal output by the PD, thereby suppressing crosstalk and improving the sensitivity of the optical transceiver module.

Abstract: A method for processing audio signals is provided comprising outputting an audio signal; receiving the output audio signal via a first receiving path as a first received audio signal; receiving the output audio signal via a second receiving path as a second received audio signal; determining an echo suppression gain based on the first received audio signal and the second received audio signal; and filtering echo suppression of the audio signal based on the first received audio signal and the echo suppression gain.

Abstract: In one form, an audio processor circuit includes a first digital signal processing circuit, a second digital signal processing circuit, and an interleaver. The first digital signal processing circuit has an input for receiving a far-end audio signal, and an output. The second digital signal processing circuit has an input for receiving a digital near-end audio signal, and an output. The interleaver has a first input coupled to the output of the first digital signal processing circuit, a second input coupled to the output of the second digital signal processing circuit, and an output for alternatively providing signals received from the first and second inputs to the output.

Abstract: According to a preferred aspect of the instant invention, there is provided a system and method that allows the user to attenuate ambient noise in speech recordings in the audio part of a video recording. The user does not need to define particular sections or samples or individual parameters. The system is automatically analyzing the input signal and in a plurality of individual steps detects the ambient noise, determines an adaptive filter, implements the filter and therewith attenuates the ambient noise accordingly.

Abstract: A system and method for sharing a designated audio signal may reproduce the designated audio signal via transducer into an acoustic space. The designated audio signal may be a recorded audio signal, an encoded audio signal, a streamed audio signal and an audio signal component of a multi-media content item. An input audio signal may be received representing a sound field in the acoustic space. One or more microphones may receive the input audio signal. A component contributed by the reproduced designated audio signal may be suppressed in the input audio signal. An echo suppressor may suppress the recaptured designated audio signal. The designated audio signal may be mixed with the suppressed input audio signal to generate an uplink audio signal. The designated audio signal mixed with the suppressed input audio signal may not include artifacts caused by the transducer, the microphone and the acoustic space.

Abstract: A personal audio device, such as a wireless telephone, includes an adaptive noise canceling (ANC) circuit that adaptively generates an anti-noise signal from a reference microphone signal and injects the anti-noise signal into the speaker or other transducer output to cause cancellation of ambient audio sounds. An error microphone is also provided proximate to the speaker to provide an error signal indicative of the effectiveness of the noise cancellation. A secondary path estimating adaptive filter is used to estimate the electro-acoustical path from the noise canceling circuit through the transducer so that source audio can be removed from the error signal. Adaptation of adaptive filters is sequenced so that update of their coefficients does not cause instability or error in the update. A level of the source audio with respect to the ambient audio can be determined to determine whether the system may generate erroneous anti-noise and/or become unstable.

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 acoustic echo cancellation (AEC) system includes a remote device, for capturing a remote captured sound, a server coupled to the remote device, and a local device coupled to the server. The server transmits the remote captured sound from the remote device to the local device. The local device receives, stores and plays the remote captured sound as a local playback sound. An echo is generated from reflection of the local playback sound. The local device captures the echo and a local sound into a local captured sound, and transmits both the remote captured sound and the local captured sound to the server. The server performs AEC on the local captured sound by using the remote captured sound from the local device and transmits the AEC processed local captured sound to the remote device.

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

Abstract: A new audio echo cancellation (AEC) approach is disclosed. To facilitate echo cancellation, the method adjusts for errors (called drift) in sampling rates for both capturing audio and playing audio. This ensures that the AEC module receives both the signals at precisely the same sampling frequency. Furthermore, the far-end signal and near-end mixed signal are time aligned to ensure that the alignment is suitable for application of AEC techniques. An additional enhancement to reduce errors utilizes a concept of native frequency. A by-product of drift compensation allows for excellent buffer control for capture/playback and buffer overflow/underflow errors from drift errors are eliminated.

Abstract: In an echo suppressor, a frequency bin component detector compares a far-end signal amplitude spectrum with a threshold value for each frequency bin to determine whether or not each frequency bin includes a frequency component. A frequency bin echo path characteristic estimator uses the far-end signal amplitude spectrum in the frequency bins determined to have a frequency component by the frequency bin component detector and the near-end input signal amplitude spectrum of corresponding frequency bins to estimate the echo path characteristics of the frequency bins. An estimated echo signal calculation-and-echo suppressor section calculates estimated echo signals based on the echo path characteristic in each frequency bin and the far-end signal amplitude spectrum to suppress the estimated echo signals from the near-end input signal amplitude spectrum.

Abstract: Embodiments of methods and devices for classifying background noise contained in an audio signal are disclosed. In one embodiment, the device includes a module for extracting from the audio signal a background noise signal, termed the noise signal. Also included is a second that calculates a first parameter, termed the temporal indicator. The temporal indicator relates to the temporal evolution of the noise signal. The second module also calculates a second parameter, termed the frequency indicator. The frequency indicator relates to the frequency spectrum of the noise signal. Finally, the device includes a third module that classifies the background noise by selecting, as a function of the calculated values of the temporal indicator and of the frequency indicator, a class of background noise from among a predefined set of classes of background noise.

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

Abstract: 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: Systems and methods are provided for echo reduction. For example, first audio data associated with a current time unit is obtained, the first audio data being generated by an audio-acquisition device of a receiver based on at least information associated with a first audio signal acquired by the audio-acquisition device and a second audio signal output from an audio-playback device of the receiver, the second audio signal being output from the audio-playback device of the receiver based on at least information associated with second audio data transmitted previously by a sender before the current time unit; third audio data transmitted by the sender during a second time unit preceding the current time unit is obtained; and fourth audio data is obtained by performing echo reduction on the first audio data based on at least information associated with the third audio data.

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: 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: 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: In one embodiment, an acoustic echo control (AEC) module receives an outgoing signal and an incoming signal, which, at various times, contains acoustic echo corresponding to the outgoing signal. The AEC module has a delay estimation block that estimates, in the time domain, the echo delay using an adaptive filtering technique. This delay estimation is used to align samples of the incoming signal having acoustic echo with the corresponding samples of the outgoing signal from which the acoustic echo originated. The AEC module determines whether or not samples of the incoming signal contain acoustic echo based on the aligned outgoing signal, and the determinations are applied to a hangover counter. The AEC module then suppresses acoustic echo in the incoming signal and adds comfort noise to the incoming signal. The amount of echo suppression performed is gradually increased or decreased based on comparisons of the counter to a hangover threshold.

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

Abstract: The active noise control device includes: a signal obtaining section that obtains an electric signal relating to the predetermined sound; a control section that adjusts an amplitude and a phase of the electric signal obtained by the signal obtaining section; a vibrating section having a diaphragm and a vibrator, the vibrator vibrating in accordance with an output from the control section. Because a sound radiated from the diaphragm toward the first region is substantially in opposite phase to that toward the second region, the control section controls the vibrator so that the diaphragm generates a sound that attenuates the predetermined sound in the first region, and causes the predetermined sound to have a desired frequency characteristic in the second region.

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 transceiver mitigates signal leakage into a receive path from a transmit path. A subtraction circuit determines a difference between a receive signal and a compensation signal to produce a compensated receive signal prior to demodulation by a demodulator. An equalizer both amplitude adjusts and phase adjusts orthogonal baseband transmit signals based on the difference from the subtraction circuit to produce the compensation signal. A digital tuning circuit determines at least one amplitude adjust coefficient to be used by the equalizer. The equalizer can have a polarity switch or a variable attenuator or a variable delay.

Abstract: The present document discloses a method and system for improving echo cancellation in the presence of non-linear distortions. A circuit attenuates a transmit signal in one or more spectral bands based on a receive signal. The distorted echo comprised within the transmit signal results from distortions incurred by the receive signal when being rendered by an electronic device. The circuit compares energy values of the receive signal in multiple spectral bands with multiple corresponding spectral band dependent distortion thresholds and determines that for a first spectral band from multiple spectral bands. A first energy value from multiple energy values exceeding a first distortion threshold from multiple distortion thresholds attenuates the transmit signal in one or more spectral bands at higher frequencies than the first spectral band with corresponding spectral band dependent attenuation factors.

Abstract: A system and method for provide a stable gain from an adaptive gain control device in a signal path. An echo canceller is also located in the signal path, and is used to provide performance information regarding losses in the signal. This performance information is fed to the automatic gain control device via a connection. The automatic gain control device thereafter uses the performance information to determine a maximum gain that might be provided based upon losses cause by echo conditions. The gain however is limited in order to provide for a stable system. The performance information includes a loss rate that includes a combination of the echo return loss and the echo return loss enhancement.

Abstract: Echo cancellation is handled using a pattern classification technique. During a training phase, the communications device is trained to learn patterns of signal inputs that correspond to certain communication modes. After numerous different patterns have been classified by mode, the classified patterns are used during real-time use of the communications device in order to determine, dynamically, the mode in which the communications device is currently operating. The communications device may then apply, to the microphone-produced signal, a suppression action.

Abstract: During playing of a streamed audio while engaging in a voice communication session, a communication device may be operable to generate a background audio during an echo cancellation operation for the voice communication session, using known information associated with the streamed audio. The streamed audio may comprise audio content that is received from one or more external sources and/or locally generated within the communication device. The known information associated with the streamed audio may comprise an echo replica corresponding to an echo associated with the streamed audio and/or a modified version of the streamed audio. In instances when a residual echo suppression is operational during the echo cancellation operation, the communication device may be operable to suppress a residual echo based on a signal level of the echo replica corresponding to the echo associated with the streamed audio and/or a signal level of the modified version of the streamed audio.

Abstract: An input/output apparatus includes a speaker, a microphone, an adaptive filter circuit and a control circuit. The speaker is configured to output audio based on a voice-receiving signal. The microphone is configured to produce a voice-transmitting signal based on gathered audio. The adaptive filter circuit is configured to produce a residual signal based on a pseudo echo signal and a amplified voice-transmitting signal. The pseudo echo signal is produced from the voice-receiving signal and a transfer coefficient which expresses audio transfer properties between the speaker and the microphone. The amplified voice-transmitting signal is produced by causing a microphone amplification circuit to amplify the voice-transmitting signal. The control circuit is configured to stop output of the residual signal from the adaptive filter circuit during at least one of the speaker and the microphone is muted.

Abstract: A method is provided which is suitable to cope with non-linear echo paths during acoustic echo cancellation in speakerphones. Non-linear paths occur particularly in hands-free operation of, e.g., a mobile phone, due to driving the amplifier and loudspeaker in the non-linear range. The idea is to combine the commonly known one microphone approach of linear acoustic echo cancellation using an adaptive filter and a post-processor together with a multiple microphone approach using beam forming which separately removes the non-linear part of the echo.

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

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

Abstract: An apparatus for providing a current loudspeaker-enclosure-microphone system description of a loudspeaker-enclosure-microphone system is provided. The apparatus has a first transformation unit for generating a plurality of wave-domain loudspeaker audio signals. Moreover, the apparatus has a second transformation unit for generating a plurality of wave-domain microphone audio signals. Furthermore, the apparatus has a system description generator for generating the current loudspeaker-enclosure-microphone system description based on the plurality of wave-domain loudspeaker audio signals, based on the plurality of wave-domain microphone audio signals, and based on a plurality of coupling values, wherein the system description generator is configured to determine each coupling value assigned to a wave-domain pair of a plurality of wave-domain pairs by determining a relation indicator indicating a relation between a loudspeaker-signal-transformation value and a microphone-signal-transformation value.