Abstract: An electronic device is provided. The electronic device includes a communication module, a sensor module, a sound input module, a sound output module, a memory including a buffer, and a processor.

Abstract: A system and method of estimating properties of a subsurface formation are described. The method includes transmitting an excitation current pulse into the formation, receiving an induced pulse used to generate input data, and performing an inversion on the input data using a lookup table based on a shape of the excitation current pulse to estimate the properties of the subsurface formation.

Abstract: Echo removal techniques are described. As part of the echo removal, a first model estimate of the echo in the received audio signal is determined using a first model and a second model estimate of the echo is determined using a second model. A first accuracy value of the first model is determined according to a model accuracy measure, and a second accuracy value of the second model is determined according to the model accuracy measure. It is then determined if the first model is more accurate than the second model based on a comparison of the first accuracy value and the second accuracy value and the second model is selectively updated based on said comparison. Echo removal is applied to the received audio signal using only the second model estimate of the echo.

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: 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: 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 system and methods for pre-configuring echo cancellers are shown and described. The system includes a storage device for storing one or more settings of the echo cancellation parameters developed during one or more previously established data connections, and an echo canceller for cancelling echo associated with one or more data connections according to echo cancellation parameters, the echo canceller initially configuring the echo cancellation parameters according to settings developed during previous data connections.

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

Abstract: An apparatus may include a noise estimation unit configured to determine a noise spectrum associated with a noise signal, and generate a filter based on the determined noise spectrum. The apparatus may also include a noise synthesis unit configured to generate a colored noise using the filter generated by the noise estimation unit. The apparatus may be incorporated in an echo canceller.

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

Abstract: The invention relates to a hearing instrument for processing an input sound to an output sound according to a user's needs. The invention further relates to a method of operating a hearing instrument and to use of a hearing instrument. The object of the present invention is to provide an alternative scheme for handling acoustic feedback in a hearing instrument.

Abstract: A communication end device of a two-way communication system is shown. The device includes an audio signal capture device for capturing local audio to be transmitted to another end device, an audio signal rendering device for playing remote audio received from the other end device, and buffers for buffering the captured and rendered audio signals. The device also includes an audio echo canceller operating to predict echo from the rendered audio signal at a calculated relative offset in the captured audio signal based on an adaptive filter, and subtract the predicted echo from the signal transmitted to the other end device The calculated relative offset that is used by the audio echo canceller for a current signal sample is adjusted if a difference between it and an adjusted relative offset of a preceding sample exceeds a threshold value.

Abstract: A wireless repeater includes an internal feedback path for adaptively cancelling an echo between an output antenna and an input antenna. The internal feedback path employs an adaptive algorithm implementing a list having a plurality of list elements. Each list element has one or more echo cancellation parameters and one or more repeater settings. The list may be pruned by employing a minimum distance between elements within the list. A method for stabilizing a wireless repeater includes obtaining an autocorrelation of a signal in a signal path of the repeater, detecting an echo in the signal path based on the autocorrelation, providing a list of elements indicating past successful echo cancellation coefficients and associated repeater settings, and adaptively adjusting an error of the echo cancellation coefficients to cancel the echo in the signal path.

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: A method includes transmitting a first signal over a network from a first communication link to a second communication link. The method further includes receiving a second signal with the first communication link from the second communication link. The method further includes canceling signal echo from the first signal present in the second signal with a digital echo canceller. The method further includes providing correction data from a memory array to the digital echo canceller during the cancellation of the signal echo. An associated apparatus is also disclosed.

Abstract: A communication device is provided that includes a speaker for outputting a ringing signal, a microphone for receiving the ringing signal, and logic configured to analyze the spectral content of the received ringing signal and adjust noise reduction parameters and echo cancellation parameters based on the analyzed spectral content of the received ringing signal.

Abstract: Disclosed herein are portable teleconferencing products that implement a doubletalk detector in a low frequency range or in a frequency range commensurate with the frequencies through which sound may be efficiently transferred between a speaker and a microphone through an enclosure. Also disclosed herein are teleconferencing systems that implement a secondary doubletalk detector, a non-presumptive doubletalk detector, a confirmatory doubletalk detector, and/or a false doubletalk detector, whereby echo cancellation coefficients may be better adapted after echo path changing events through the use of accelerated coefficient adaptation or half-duplex operation until adaptation is restored.

Abstract: A method is set forth for reducing the total acoustic echo cancellation convergence time for all look directions in a microphone array based full-duplex system. The method is based on capturing the loudspeaker signal due to the first far-end speech bursts when the conferencing system is first used, as well as the corresponding loudspeaker feedback signals in the individual microphones. The captured signals are then used for consecutive adaptation of the acoustic echo canceller on all echo paths corresponding to all look directions of the beamformer, thereby training the AEC. This training process can be executed concurrently with normal phone operation, for example, as a background process that utilizes available processing cycles.

Abstract: Echo canceling method comprising the steps: a) receiving a first signal and a second signal, said second signal comprising an echo of the first signal, b) generating an echo replica signal by filtering the first signal with a finite impulse response filter, said finite impulse filter using a filter coefficient vector for generating the echo replica signal, generating an echo cancelled signal by subtracting the echo replica signal from the second signal, d) determining a measure of interference within the second signal, e) determining a step vector depending on said measure of interference, wherein increasing the measure of interference continuously reduces a size of the step vector, and f) updating the filter coefficient vector by the step vector.

Abstract: A method and apparatus for reducing crosstalk in a multi-channel communication system is disclosed. In one embodiment, outgoing signals in a multi-channel environment are manipulated into a transform domain, such as the frequency domain. Thereafter, the signals may be combined and modified based on a weighting variable to create a cancellation signal. Combined processing greatly reduces system complexity and increases processing speed. After processing in the transform domain, the cancellation signal undergoes further processing to return the cancellation signal into the time domain. The cancellation signal may then be combined with received signals to cancel crosstalk or echo. A method and apparatus for crosstalk cancellation in the analog domain and digital domain is 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: 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 apparatus for canceling a signal is disclosed. The apparatus may include an identification code (ID code) generator configured to generate an ID code. The apparatus may also include an ID code injector configured to inject the ID code into at least one of the signal and a processed signal to produce a convolved signal. The processed signal is resulted from a processing of the signal. The apparatus may further include an ID code detector configured to detect at least one of the convolved signal, a transformed signal, and a transformation of the convolved signal. The transformed signal is resulted from the transformation of the convolved signal. The apparatus may further include an arithmetic function configured to remove at least one of the convolved signal and the transformed signal.

Abstract: The present invention relates to an audio communication system and method with improved acoustic characteristics. The present invention utilizes the Haas effect to simulate a sound picture that gives a better subjective impression of stereo sound compared to the objective stereo image. It provides a system and a method for presenting the stereo image in an optimal way for an associated echo canceller without compromising with the subjective perception of stereo.

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: An affine projection or like algorithm is used to process a non-stationary signal. The affine projection algorithm creates an inverse matrix and includes a factor ? to avoid numerical instability in the inverse matrix. The factor ? is adaptively adjusted according to the characteristics of the signal.

Abstract: A communication system having an echo canceller is disclosed. One embodiment of the echo canceller includes an adaptive filter used to provide an estimate of reflected echo which is removed from the send signal. The echo canceller may also include a near-end talker signal detector which may be used to prevent the adaptive filter from adapting when a near-end talker signal is present. The echo canceller may also include a nonlinear processor used to further reduce any residual echo and to preserve background noise. The echo canceller may also include a monitor and control unit which may be used to monitor the filter coefficients and gain of the adaptive filter to maintain stability of the echo canceller, estimate pure delay, detect a tone, and inject a training signal. The echo canceller may also include a nonadaptive filter used to reduce the length of the adaptive filter.

Abstract: A method of suppressing an echo signal in a communication path such as acoustic coupling between a speaker and a microphone in a telephone device handset includes the steps of monitoring a transmitted signal supplied to the communication path to determine an attribute thereof; generating a leaky mask based on the determined attribute and an attribute of a received echo signal; and partially cancelling the received echo signal using the leaky mask.

Abstract: Providing multi-path echo cancellation comprises providing a main echo signal corresponding to a main echo section of an echo response and providing a plurality of tail echo signals, each of the plurality of tail echo signals corresponding to respective tail echo sections of the echo response. Next, providing a multi-path echo canceller comprises summing the main echo signal and the plurality of tail echo signals to provide a synthesized echo.

Abstract: An apparatus and method for echo cancellation that detects and tracks the active regions of the echo channel impulse response rather than the entire echo channel impulse response and uses this sparse model to cancel the echo component of the received signal. Two sets of filter coefficients are used, one to remove and track the active regions and one to detect the active regions.

Abstract: A distributed echo cancelling architecture is provided where echo-cancelling functions are performed at locations remote from devices receiving signals with echoes. The echo cancelling functions use a reference signal, which has been corrupted with the echoes at the devices, for echo cancellation. As echo canceller resources are located at a central system and not at each individual device, the echo canceller resources can therefore be shared between the devices.

Abstract: A device has a first echo canceller having a first bank of coefficients to be applied to a reference signal to produce a first echo prediction signal, and a first arithmetic unit to combine the first echo prediction signal with the echo-bearing signal to produce a first echo-cancelled signal. A second echo canceller receives the reference signal and the first echo-canceled signal, and has a second bank of coefficients to be applied to the reference signal to produce a second echo prediction signal. The second bank of coefficients is adapted to improve echo cancellation in the second echo canceller. A second arithmetic unit combines the first echo-cancelled signal with the second echo-prediction signal to produce a second echo-cancelled signal.

Abstract: In a microphone signal, the signal component corresponding to, e.g., echo is suppressed using an echo control scheme that estimates the spectral envelope of the echo signal, without having to estimate the waveform for the echo signal. In one embodiment, the input signal (to be applied to a loudspeaker) and the microphone signal are spectrally decomposed into multiple subbands, where echo suppression processing is independently performed on each subband. The echo control of the present invention can be implemented with substantially reduced (1) computational complexity and (2) phase sensitivity, as compared to traditional acoustic echo cancellation, in which the waveform for the echo signal is estimated.

Abstract: An echo canceller circuit for use in an echo canceller system is set forth that provides sensitive non-linear echo path response detection. The echo canceller circuit comprises a first digital filter having non-adaptive tap coefficients to simulate an echo response occurring during a call. A second digital filter having adaptive tap coefficients to simulate an echo response occurring during the call is also used. The adaptive tap coefficients of the second digital filter are updated over the duration of the call. A coefficient transfer controller is disposed in the echo canceller circuit to transfer the adaptive tap coefficients of the second digital filter to replace the tap coefficients of the first digital filter when a set of one or more transfer conditions is met. A non-linear echo path response detector is provided. The non-linear echo path detector is responsive to one or more parameters of the first and second digital filters to detect a non-linear echo path condition.

Abstract: An apparatus and technique for determining whether a telephone loop is capable of carrying a Digital Subscriber Line (DSL) signal. Time domain reflected (TDR) signals that are echoes of transmit impulse signals are processed using time domain echo averaging and echo enhancement using second-order statistics. Each technique allows clearer visualization of the TDR echo signal by reducing the noise and distortion present in the TDR signals. These statistical signal processing techniques allow determination of telephone loop characteristics such as loop length once the TDR echo signal amplitude and time index are calculated.

Abstract: A digital echo cancellation device is provided. The digital echo cancellation device used for a high speed bidirectional communication system includes an adaptive beamformer in the form of a plurality of finite impulse response (FIR) filters for estimating an input receiving signal, the adaptive beamformer for estimating a front part, which rapidly changes in an echo path impulse response, by adaptively estimating the input receiving signal and an orthogonalized infinite impulse response (IIR) filter for receiving the estimated signal output from the adaptive beamformer and estimating a hind part of the echo path impulse response on the basis of an IIR. According to the digital echo cancellation device, the amount of calculation and the amount of required memory is significantly reduced, convergence speed is high, and the stability of the output of the filter is improved since the impulse response of the echo path is estimated by only several tens of taps.

Abstract: A method and apparatus for reducing crosstalk in a multi-channel communication system is disclosed. In one embodiment, outgoing signals in a multi-channel environment are manipulated into a transform domain, such as the frequency domain. Thereafter, the signals may be combined and modified based on a weighting variable to create a cancellation signal. Combined processing greatly reduces system complexity and increases processing speed. After processing in the transform domain, the cancellation signal undergoes further processing to return the cancellation signal into the time domain. The cancellation signal may then be combined with received signals to cancel crosstalk or echo. A method and apparatus for crosstalk cancellation in the analog domain and digital domain is also disclosed.

Abstract: A method of converging a step size control for an adaptive filter of a communication channel including: (1) initializing a nominal step size value and a penalty point value; (2) combining the nominal step size value and the penalty point value to generate a step size value; and (3) dynamically changing the step size value in response to a characteristic measure of a quality of the communication channel. The step size value is changed by adjusting the nominal step size value, the penalty point value, or both. In a preferred embodiment the penalty point value is adjusted dependant on: (1) a tone originating from the far end (2) full convergence (3) the power level of a residual error signal (4) the channel's near-end background noise and/or (5) weak double-talk in the communication channel. The nominal step size value is adjusted when an achieved initial combined loss is about 15 dB or greater and is reset by divergence.

Abstract: An echo canceler includes an adaptable filter that is updated during double-talk and far-end talk. The echo canceler receives a microphone signal that includes a near-end signal and echoes resulting from a far-end signal. The echo canceler whitens the far-end signal and decorrelates the microphone signal. The whitening and decorrelating is done as a preprocessing step for the one-unit blind source separation algorithm. A criterion of measure is used to emphasize the statistical independence of the echo canceled signal and the far-end signal. The echo cancellation problem is cast as a one-unit blind source separation problem. A one-unit blind source separation algorithm such as negentropy maximization can be used to extract the echo canceled signal.

Abstract: An apparatus and method for echo cancellation that detects and tracks the active regions of the echo channel impulse response rather than the entire echo channel impulse response and uses this sparse model to cancel the echo component of the received signal. Two sets of filter coefficients are used, one to remove and track the active regions and one to detect the active regions.

Abstract: Two signals, which are mutually correlated, are subjected to a principal component analysis and converted into two signals being put in an orthogonal relation, thereby generating two signals being non-correlated. Those two signals are reproduced by speakers, and the voice generated from the speakers are collected by microphones. The cross spectra of a signal as the result of subtracting an echo canceling signal from a voice collected by each microphone, and a voice before it is generated from the speaker, are obtained. Those cross spectra are ensemble-averaged for a predetermined period of time, and inverse Fourier transformed, thereby producing impulse response estimation errors of each filter. Impulse responses of those filters are updated so as to cancel those impulse response estimation errors.

Abstract: A process is described which generates matrix coefficients using zero-lag auto and cross-correlations from signals commonly found in echo cancellers. Double-talk and path changes are then detected using matrix operations such as determinants, eigendecompositions, or singular value decompositions (SVDs). In a preferred embodiment, the determinant of the correlation-based matrix is compared against predetermined threshold values.

Abstract: An echo canceler for generating an echo replica and subtracting the echo replica from a local input signal to create a residual signal for outgoing transmission. An echo canceler has adaptive filters that have different computational accuracy generating the echo replica. An evaluator coupled to the adaptive filters groups coefficients into each of segments, and evaluating a computational accuracy information according to the coefficients for the segments. A computational accuracy controller in the evaluator assigns the local input signal to the adaptive filters, and adjusts the coefficients according to the computational accuracy information.

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: A communication transmitting and receiving system in which the effects of near-end echo and near-end crosstalk signal from the communication medium are mitigated by adaptively reproducing the near-end echo and near-end crosstalk signal, which is then subtracted from the received signal. Filter coefficients for a Finite Impulse Response filter are adaptively generated to reproduce the near-end echo and near-end crosstalk. The filter coefficients are regenerated for the Finite Impulse Response filter in an adaptive correlator at the arrival of each received signal and whereby each new filter coefficient is a weighted sum of a previous coefficient and one received signal multiplied by a time delayed version of one transmitted signal.

Abstract: Long distance telephone connections must commonly employ echo cancellers to eliminate irritating echo. The normal practice is to additionally employ non-linear processors (NLPs) as an integral component of an echo canceller in order to eliminate small residual echo which remains after initial echo cancellation. While such devices do indeed remove remaining residual echo, they also may introduce additional problems, such as suppression of the background signal. However, in some cases there is no need to deploy a potentially problematic non-linear processor. If the overall route delay of the telephone connection is not excessive, the participants do not notice the residual echo. Therefore, a determination is made of the end to end delay of the overall telephone connection. If the determined delay is not excessive, non-linear processing is not applied or is applied less aggressively. This results in a clearer telephone call, free from non-linear processor problems.

Abstract: A fast convolution method applicable to convolving a signal (indicative of an n-dimensional pattern, where n is greater than or equal to two) with a smooth kernel that can be approximated by a separated-spline kernel, and a system configured to perform such method using software or signal processing circuitry. Unlike Fourier-based convolution methods which require on the order of N log N arithmetic operations for a signal of length N, the method of the invention requires only on the order of N arithmetic operations to do so. Unlike wavelet-based convolution approximations (which typically also require more arithmetic operations than are required in accordance with the invention to convolve the same signal), the method of the invention is exact for convolution kernels which are spline kernels.

Abstract: Real and imaginary components of the discrete Fourier transform (DFT) of the residual echo in an acoustic echo cancellation system are separately estimated using mapping functions derived without an assumption of statistical independence between separate components of a near-end microphone input signal. The residual echo estimator calculates the real and imaginary components using an approximation of the microphone input signal without the echo. A different approximation is used depending on whether the near-end talker and/or the far-end talker are talking.

Abstract: A method and apparatus that performs specific automatic gain control in an echo canceller (16). The disclosed method and apparatus utilize a low complexity algorithm that performs automatic gain control to achieve greater speed of convergence of the echo canceller (16) and an increased ability to distinguish between near-end speech and echo through the use of specific automatic gain control. A double talk threshold of the echo canceller is adjusted by using, in part, computed gains of inbound and outbound signals internally within the echo canceller in order to adjust the echo canceller double talk threshold.