Abstract: An acoustic echo canceller (AEC) uses a simulated echo from an adaptive filter to cancel an actual echo from a real environment. The echo cancellation is optimal when the adaptive filter response accurately matches the environment response. This can be achieved by changing coefficients of the filter in a step-wise fashion until the filter is converged on a configuration that provides optimal results. The convergence of the filter can be negatively affected by interference due to a double talk event or by an event in which the environment is changed. The disclosed AEC is configured to respond to these disruptive events by adjusting a step-size of the adaptive algorithm used to change the coefficients.

Abstract: An apparatus includes one or more processors and logic encoded in one or more non-transitory media for execution by the one or more processors and when executed operable to receive images from a set of cameras that are positioned in a plurality of levels around one or more objects, wherein the set of cameras are positioned at a same orientation and each image is captured by a rolling shutter and associated with a position and orientation of a corresponding camera from the set of cameras. The logic is further operable to, for each level of cameras generate a three-dimensional pointcloud based on the position and orientation of corresponding cameras and matched extracted features. The logic is further operable to merge the three-dimensional pointcloud for each level of the cameras to form a three-dimensional volumetric model of one or more objects.

Abstract: An echo cancellation device and an echo cancellation method thereof applied in a communication device are provided. The echo cancellation device includes an echo canceller and a combine circuit. The echo canceller obtains a plurality of delayed signals from a local signal of the communication device, and the delayed signals are divided into a plurality of delayed signal groups. The echo canceller selectively ignores at least one of the delayed signal groups, and the echo canceller generates an echo cancellation signal with the others of the delayed signal groups. The combine circuit is coupled to an interface circuit of the communication device to receive a received signal. The combine circuit cancels an echo component of the received signal with the echo cancellation signal to generate a cancelled signal.

Abstract: By performing both noise cancellation and echo cancellation, a high-quality main voice signal is generated. A voice input/output apparatus includes a noise acquirer that is arranged toward an outside of a body of a user and acquires external noise arriving from the outside of the user, a voice output unit that accepts an input of a voice signal and outputs a voice to an ear canal of the user, a main voice acquirer that acquires a mixed voice, in which the external noise, the output voice, and a main voice of the user transmitted from a vocal cord of the user through the ear canal are mixed, and outputs a mixed voice signal, a noise canceler that processes the mixed voice signal using a noise signal based on the external noise, and an echo canceler that processes the mixed voice signal using the voice signal.

Abstract: A method of noise cancellation comprises providing a first estimated output signal based on an input signal and a first prior far-end channel filter and a second prior far-end channel filter corresponding to a prior frame, updating a first current far-end channel filter corresponding to a current frame according to the first estimated output signal and the first prior far-end channel filter, providing a second estimated output signal based on the input signal, the first current far-end channel filter, and the second prior far-end channel filter, updating a second current far-end channel filter corresponding to the current frame according to the second estimated output signal and the second prior far-end channel filter, and providing a resultant signal based on the input signal, the first current far-end channel filter, and the second current far-end channel filter.

Abstract: A system that performs wall detection, range estimation, and/or corner detection to determine a position of a device relative to acoustically reflective surfaces. The device generates output audio using a loudspeaker, generates microphone audio data using a microphone array, performs beamforming to generate directional audio data and then generates impulse response data for each of a plurality of directions. The device may detect a peak in the impulse response data and determine a distance and/or direction to a reflective surface based on the peak. Based on a number of reflected surfaces and/or direction(s) of the reflected surfaces detected by the device, the device may classify the different directions and estimate where it is in the room, such as whether the device is in a corner, along one wall, or in an open area. By knowing its position relative to the room surfaces, the device may improve sound equalization.

Abstract: Acoustic echo cancellation (AEC) processing may be improved by performing echo cancellation using a combined multi-channel reference signal. Two or more reference signals, such as a left and right channel of a stereo source, may be combined and provided to an AEC block configured to receive the combined signal and perform AEC processing using the combined signal. The AEC block may include an adaptive filter that performs operations that cause pre-whitening of the combined reference signal and de-correlation of the individual channels within the combined reference signal. The pre-whitening of the signal flattens the spectrum of the combined reference signal, which may improve convergence speed of the AEC processing in cancelling the echo. The de-correlating of the signal cancels inter-channel correlation between the multiple channels, which may improve convergence speed of the AEC processing in cancelling the echo.

Abstract: A device that determines an echo latency estimate by combining reference signals. The device may determine the echo latency corresponding to an amount of time between reference signals being sent to transmitters and input data corresponding to the reference signals being received. The device may generate a combined reference signal by adding (or filtering) each of the reference signals. The device may then compare the combined reference signal to input audio data received from a microphone or receiving device. The device may detect a highest peak, determine if there are any earlier significant peaks and estimate the echo latency based on the earliest significant peak. This technique is not limited to audio data and may be used for signal matching using any system that includes multiple transmitters and receivers (e.g., Radar, Sonar, etc.).

Abstract: An acoustic signal processing system and method. In accordance with an embodiment, the acoustic signal processing system includes an adaptive filter that filters a signal from a frequency band reservation module and generates a filter signal that is received by a subtractor. The subtractor generates an error signal that is used by a double-talk indicator module to generate a control signal that indicates the presence of double-talk.

Abstract: A device may perform audio signal collision detection configured to receive first audio information from a first device and receive second audio information from a second device over a communications network. The device may be further configured to identify a collision of the first audio information and the second audio information, and to address the collision by adjusting at least one of an audio attenuation level and a communications latency of the second audio information responsive to identification of the collision, and store the operational parameters to an adjustment profile system.

Abstract: Embodiments of the present invention exploit redundancy of succeeding FFT spectra and use this redundancy for computing interpolated temporal supporting points. An analysis filter bank converts overlapped sequences of an audio (ex. loudspeaker) signal from a time domain to a frequency domain to obtain a time series of short-time loudspeaker spectra. An interpolator temporally interpolates this time series. The interpolation is fed to an echo canceller, which computes an estimated echo spectrum. A microphone analysis filter bank converts overlapped sequences of an audio microphone signal from the time domain to the frequency domain to obtain a time series of short-time microphone spectra. The estimated echo spectrum is subtracted from the microphone spectrum. Further signal enhancement (filtration) may be applied. A synthesis filter bank converts the filtered microphone spectra to the time domain to generate an echo compensated audio microphone signal.

Abstract: Audio from a near-end that has been acoustically coupled at the far-end and returned to the near-end unit is detected and suppressed at the near-end of a conference. First and second energy outputs for separate bands are determined for the near-end audio being sent from the near-end unit and for the far-end audio being received at the near-end unit. The near-end unit compares the first and second energy outputs to one another for each of the bands over a time delay range and detects the return of the sent near-end audio in the received far-end audio based on the comparison. The comparison can use a cross-correlation to find an estimated time delay used for further analysis of the near and far-end energies. The near-end unit suppresses any detected return by muting or reducing what far-end audio is output at its loudspeaker.

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

Abstract: A howling canceller applied to an acoustic system having a speaker and a microphone comprises: a filter insertion unit for inserting a notch filter at a frequency of an audio signal picked up by the microphone; a setting unit for setting the insertion time of the notch filter on the basis of the frequency at which the notch filter is inserted; and a releasing unit for, when the insertion time set by the setting unit has elapsed, releasing the notch filter, the insertion time of which has elapsed. The setting unit sets the insertion time of the notch filter to be shorter as the frequency at which the notch filter is inserted increases.

Abstract: A method of echo cancellation, particularly for use in high definition applications, splits an input signal and reference signal into M single-sided sub-band. The subbanded signals are downsampled at a downsampling rate N, where N?M, adaptively filtered, and recombined to produce an output signal. The sub-bands are preferably oversampled such that N<M. The use of oversampling and single-sided sub-banding reduces complexity and avoids aliasing problems.

Abstract: A method and system for improving a perceived duplexity of handsfree telephone applications is disclosed. An echo suppression circuit for a device comprising loudspeaker and microphone is described. A circuit attenuates a subband of a transmit signal, wherein the transmit signal is captured by the microphone and wherein the transmit signal comprises an echo of a far-end signal rendered by the loudspeaker and a near-end signal. The attenuation circuit further determines a subband far-end indicator of a voice activity in the far-end signal; determines a subband near-end indicator of a voice activity of the near-end signal; determines a subband masking weight; determines a subband attenuation for the transmit signal in the subband; and attenuates the subband of the transmit signal using the determined subband attenuation.

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: An audio system with at least one audio channel may include a digital audio processor in which at least one digital filter is implemented for each channel. The digital filter of each channel may include: an analysis filter bank configured to receive a broad-band input audio signal and divide the input audio signal into a plurality of sub-bands, a sub-band filter for each sub-band, and a synthesis filter bank configured to receive the filtered sub-band signals and combine them for providing a broad-band output audio signal. A delay is associated with each sub-band signal, the delay of one of the sub-band signals being applied to the broad-band input audio signal upstream of the analysis filter bank and the residual delays being applied to the remaining sub-band signals downstream of the analysis filter bank.

Abstract: A noise canceling headphone includes a microphone provided in a headphone housing, collecting external sound, a canceling signal generating circuit generating a canceling signal in response to a signal output from the microphone which cancels external sound passing through the housing without being attenuated by a sound insulating characteristic of the headphone, a compensating signal generating circuit generating a compensating signal that compensates for the external sound attenuated by the sound insulating characteristic of the headphone, an adding circuit that adds a musical signal input from the exterior to the compensating signal, and a speaker unit that outputs an output signal from the adding circuit and the canceling signal.

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 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: Audio from a near-end that has been acoustically coupled at the far-end and returned to the near-end unit is detected and suppressed at the near-end of a conference. First and second energy outputs for separate bands are determined for the near-end audio being sent from the near-end unit and for the far-end audio being received at the near-end unit. The near-end unit compares the first and second energy outputs to one another for each of the bands over a time delay range and detects the return of the sent near-end audio in the received far-end audio based on the comparison. The comparison can use a cross-correlation to find an estimated time delay used for further analysis of the near and far-end energies. The near-end unit suppresses any detected return by muting or reducing what far-end audio is output at its loudspeaker.

Abstract: Regardless of the presence of echo or uplink speech, a microphone signal is received at a small filter and this small filter is functionally and/or physically separate from an adaptive echo canceller filter. The signal is applied to the small filter and an error signal is determined from the signal utilizing the small filter. The small filter continuously adapts the received signals. A first adaptation factor is determined based at least upon the error signal and the microphone signal according to a first signal analysis approach and a second adaptation factor is determined based at least upon the microphone signal according to a second signal analysis approach. The first adaption factor is compared to the second adaptation factor and one of the first adaptation factor or the second adaptation factor is selected based upon at least one predetermined criteria. The selected adaptation factor is applied to the echo canceller filer to control the convergence of the echo canceller filter.

Abstract: The present invention relates to design and implementation of low complexity adaptive echo and NEXT cancellers in multi-channel data transmission systems. In this invention, a highly efficient weight update scheme is proposed to reduce the computational cost of the weight update part in adaptive echo and NEXT cancellers. Based on the proposed scheme, the hardware complexity of the weight update part can be further reduced by applying the word-length reduction technique. The proposed scheme is general and suitable for real applications such as design of a low complexity transceiver in 10GBase-T. Different with prior work, this invention considers the complexity reduction in weight update part of the adaptive filters such that the overall complexity of these adaptive cancellers can be significantly reduced.

Abstract: A method of acoustic echo cancellation in the VoIP terminal using processing of the far-end signal with the digital adaptive filter in order to obtain the echo estimate that is subtracted from the microphone signal in which the far-end signal, before it is converted to the analog form and passed to the loudspeaker (4), is marked by embedding an encoded digital signature obtained from the signature generator (14) and then detection of the digital signature is performed in the signal collected by the microphone (7) and converted to digital form, depending on the result of the digital signature detection, adaptation of the digital adaptive filter (9) is resumed or stopped.

Abstract: Methods and systems for echo modulation are described. In one embodiment, intensities of a plurality of values in multiple windows of an audio signal may be obtained. The windows may be subject to a periodic boundary condition. A plurality of echo values may be calculated for each of the respective windows. The audio signal may be altered in one or more of the windows using a windowing function and echo values. Additional methods and systems are disclosed.

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

Abstract: An audio signal decorrelator for deriving an output audio signal from an input audio signal has a frequency analyzer for extracting from the input audio signal a first partial signal descriptive of an audio content in a first audio frequency range and a second partial signal descriptive of an audio content in a second audio frequency range having higher frequencies compared to the second audio frequency range. A partial signal modifier modifies the first and second partial signals, to obtain first and second processed partial signals, so that a modulation amplitude of a time variant phase shift or time variant delay applied to the first partial signal is higher than that applied to the second partial signal, or for modifying only the first partial signal. A signal combiner combines the first and second processed partial signals, or combines the first processed partial signal and the second partial signal, to obtain an output audio signal.

Abstract: Different sampling rates between a playout unit and a capture unit are compensated for via a system, method and computer program product. The playout unit receives samples from a computational unit, and the capture unit sends samples to the computational unit. A playout FIFO buffer operates in a playout time domain, and a capture FIFO buffer operates in a capture time domain. The computational unit is synchronized to a common clock. A first relationship is calculated between the common clock and a playout fifo buffer read pointer, and a second relationship is calculated between the common clock and a capture FIFO buffer write pointer. For each sample in the playout time domain a corresponding sample in the samples from said computational unit is found and sent to the playout FIFO buffer. For each sample in the common clock time domain the corresponding sample in the capture time domain is found and sent to the computational unit.

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

Abstract: A sound source model storage section stores a sound source model that represents an audio signal emitted from a sound source in the form of a probability density function. An observation signal, which is obtained by collecting the audio signal, is converted into a plurality of frequency-specific observation signals each corresponding to one of a plurality of frequency bands. Then, a dereverberation filter corresponding to each frequency band is estimated by using the frequency-specific observation signal for the frequency band on the basis of the sound source model and a reverberation model that represents a relationship for each frequency band among the audio signal, the observation signal and the dereverberation filter. A frequency-specific target signal corresponding to each frequency band is determined by applying the dereverberation filter for the frequency band to the frequency-specific observation signal for the frequency band, and the resulting frequency-specific target signals are integrated.

Abstract: Embodiments of the present disclosure generally include a method and apparatus for minimizing acoustic echo in video conference. In one embodiment, a computer implemented method for minimizing acoustic echo in video conferencing comprises determining position information wherein the position information is derivable from a video stream and processing the position information using a computer to identify at least one direction wherein the one direction minimizes acoustic echo.

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: 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: A combined sidetone and hybrid balance apparatus and method of operating same are disclosed. An embodiment of the present invention may provide both a hybrid balance mode of operation and a sidetone generation mode of operation within a single integrated circuit device. The functionality provided by an embodiment of the present invention may be used in both Internet protocol (IP)-based telephones and residential gateways, and may be incorporated within the IP telephone chip used in such devices. An embodiment of the present invention may be used to reduce processor demand and the cost of materials, permitting greater IP telephone or residential gateway functionality at lower cost.

Abstract: Embodiments of a multi-channel audio communication device and methods for reducing echoes are generally described herein. Other embodiments may be described and claimed. In some embodiments, echo-reduction circuitry may insert training signals into digital audio signals at or below a noise floor and in a non-audible portion of the frequency spectrum based on a spectral mask. Training signals may be generated from spectrum estimates and training sequences. An adaptive filter may generate echo-cancellation signals using filter coefficients generated fawn the training sequences and return-path signals. The echo cancellation signals may remove echo signals from the return-path signals.

Abstract: Echo cancellation for audio/video conferencing uses an inaudible tone added at an originating conference site to the transmitted voice signal, for echo cancellation. The tone, upon receipt with the voice signal at a remote conferencing site and retransmission back to the originating site, is used to determine both the round trip delay of the conferencing link and the acoustic characteristics of the remote site, so as generate an echo cancellation signal for better audio performance.

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 audio signal is processed by transforming the signal into a frequency domain representation having a plurality of frequency subbands. A decorrelated signal is derived from the frequency domain representation using a phase rotation.

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

Abstract: Systems and methods for envelope-based acoustic echo cancellation in a communication device are provided. In exemplary embodiments, a primary acoustic signal is received via a microphone of the communication device, and a far-end signal is received via a receiver. Frequency analysis is performed on the primary acoustic signal and the far-end acoustic signal to obtain frequency sub-bands. An echo gain mask based on magnitude envelopes of the primary and far-end acoustic signals for each frequency sub-band is generated. A noise gain mask based on at least the primary acoustic signal for each frequency sub-band may also be generated. A combination of the echo gain mask and noise gain mask may then be applied to the primary acoustic signal to generate a masked signal. The masked signal is then output.

Abstract: A method of detecting double talk condition in hands free communication devices is disclosed. In general, the method in accordance with the teachings of this invention detects double talk conditions based on inherent frequency response differences between the transducers used and acoustical effect on the spectrum of the returned echo signal. An input signal from a far-end talker and an input signal from the output from an echo canceler are received by the detector. K spectral subbands are created for each input signal. From this K subbands q subbands are selected based on inherent frequency differences between the far-end transducer and a near-end transducer. The spectral echo residual power is estimated at each subband. The estimated spectral echo power and the output signal from the echo canceler for a selected subband are compared to a predetermined threshold. Based on this comparison, it is determined whether double talk conditions exist based on the comparison.

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: Disclosed herein is an echo canceller for use in a sound reinforcement communication system configured to carry out a sound reinforcement communication by utilizing a speaker and a microphone, the echo canceller including: an adaptive filter section configured to adaptively identify an impulse response of a feedback path formed by an acoustic coupling or the like between the speaker and the microphone to estimate an echo component in the feedback path from an input signal to the feedback path, and subtracting the echo component thus estimated from an output signal from the feedback path; and an echo suppressing section configured to execute echo suppressing processing for an output signal from the adaptive filter section.

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 signal processing system detects reverberation. The system may suppress the reverberation and improve signal quality. The system analyzes frequency bands of an input signal to determine whether reverberation characteristics are present. When reverberation is detected, the system may attenuate the reverberant frequency band to reduce or eliminate the reverberation.

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: An echo canceller, in which an adder adds a pseudo echo generated by an adaptive filter to an echo, so that the echo and the pseudo echo are canceled to each other, includes: a subband filter (8, 14) and a sampling converter (9, 15) for downsampling an audio signal; a selector switch (10, 16) which appropriately switches between a side performing the downsampling and another side performing fullband echo cancellation; and a tap arrangement controller (19) which optimizes a tap arrangement by using coefficients which are updated using a downsampled signal by the adaptive filter (11), the tap arrangement controller outputting the optimized tap arrangement to the adaptive filter (11) at time of the fullband echo cancellation.