Abstract: Various embodiments relate to detecting collisions on a communication network. A method may include transmitting a first signal to a shared bus. The method may also include observing a second signal at the shared bus during the transmitting. Further, the method may include detecting a collision on the shared bus in response to an amplitude of the second signal being one of greater than a first threshold and less than a second threshold.

Abstract: According to an aspect, a method for multi-channel echo cancellation includes receiving a microphone signal and a multi-channel loudspeaker driving signal. The multi-channel loudspeaker driving signal includes a first driving signal that drives a first loudspeaker, and a second driving signal that drives a second loudspeaker. The first driving signal is substantially the same as second driving signal. The microphone signal includes a near-end signal with echo. The method includes determining a unique solution for acoustic transfer functions for a present acoustic scenario based on the microphone signal and the multi-channel loudspeaker driving signal. The acoustic transfer functions include first and second acoustic transfer function. The unique solution is determined based on time-frequency transforms of observations from the present acoustic scenario and at least one previous acoustic scenario.

Abstract: Provided is a method for eliminating an echo. The method includes: obtaining a mixed signal of a mixture of a user's voice given out from the user and the echo given out from a loudspeaker, and obtaining a sound signal given out from the loudspeaker based on the volume of the sound given out from the loudspeaker, the sound signal being used as a reference signal; obtaining a gain coefficient of the reference signal based on comparing the mixed signal with the reference signal, and obtaining a gain signal of the reference signal through the gain coefficient; obtaining a compensation signal of the reference signal based on the gain signal and a corresponding preset compensation coefficient; inverting the compensation signal; and combining the inverted compensation signal with the mixed signal, so as to eliminate the echo given out from the loudspeaker. A system for eliminating the echo is also provided.

Abstract: An electronic device, an echo signal cancelling method thereof and a non-transitory computer readable recording medium is provided. The electronic device according to an exemplary embodiment includes a speaker configured to output a sound corresponding to a reference signal, a microphone configured to generate a microphone signal by obtaining a received sound, and a filter configured to cancel an echo signal of the reference signal from the microphone signal. In addition, the filter includes a first filter configured to estimate an echo signal of the reference signal and cancel the estimated echo signal from the microphone signal, and a second filter configured to generate an adaptive gain to cancel a residual echo from the microphone signal in which the estimated echo signal is canceled, and generate an output signal by using the generated adaptive gain and the microphone signal in which the estimated echo signal is canceled.

Abstract: Disclosed are an apparatus and a method of cancelling an echo signal flowing in a microphone of an electronic device.

Abstract: A voice recognition system in a vehicle includes: a first microphone mounted in the vehicle that collects voice data of an occupant of the vehicle; a second microphone provided in a mobile device of the occupant that collects voice data of the occupant; and a voice recognition device connected to the mobile device through local wireless communication including a noise elimination portion eliminating noise in the voice data collected by the first microphone or the second microphone and a voice recognition portion performing voice recognition using the voice data from which noise is eliminated by the noise elimination portion.

Abstract: An input ambisonic audio signal includes multiple channels, each of which is made up of audio data representing sound captured by an ambisonic microphone. A remote audio signal made up of audio data representing sound captured by remote meeting equipment is output by a local loudspeaker. Acoustic echo cancellation is performed on the input ambisonic audio signal by removing the remote audio signal from the input ambisonic audio signal. The acoustic echo cancellation may be performed on ambisonic A-format or B-format encoded audio data, or on an output encoding generated from the B-format encoded audio data. Comfort noise may be generated based on spectral and spatial characteristics of noise in the input audio data, for insertion into the input signal during acoustic echo cancellation. Automatic gain control may be performed across the multiple channels of the input audio signal.

Abstract: An audio-based system may perform echo cancellation using first and second adaptive filters. An adaptation controller may be configured to control whether and/or how the filter coefficients of the first adaptive filter are updated, based on detected filter divergence, echo path changes, and/or presence of near-end user voice. The parameters of the second adaptive filter may be copied from the first adaptive filter under certain conditions that indicate whether copying the parameters would be likely to improve echo suppression.

Abstract: An echo cancellation apparatus cancels an echo signal included in a second speech signal including the echo signal of a first speech signal, and includes: a first echo replica generating unit generating a first echo replica by processing the first speech signal through a filter whose characteristic is equalized with a acoustic transfer function of amplified sound; a first echo cancelling unit generating and providing a first echo-cancelled signal by subtracting from the second speech signal the first echo replica generated by the first echo replica generating unit; and a filter updating unit generating a nonlinear-converted echo-cancelled signal by performing nonlinear conversion on the first echo-cancelled signal generated by the first echo cancelling unit, and updating the filter characteristic thereby equalizing the filter characteristic with the acoustic transfer function, using the nonlinear-converted echo-cancelled signal, the first speech signal, and a norm of the first speech signal.

Abstract: The present invention relates to a method and arrangement for an improved noise canceller in a speech encoder. Sound signals are captured at a primary microphone in conjunction with a reference microphone. An adaptive shadow filter is adapted to the correlation between the signals captured at the primary and reference microphones. Further, a diffuse-noise-field detector is introduced which detects the presence of diffuse noise. When the diffuse-noise-field detector detects diffuse noise, the filter coefficients of the adapted shadow filter is used by a primary filter to cancel the diffuse noise at the signal captured by the primary microphone. Since the filter coefficients of the adapted shadow filter only is used for cancellation when diffuse noise is solely detected, cancellation of the speech signal is avoided.

Abstract: A method and an audio processing system determine a system parameter, e.g. step size, in an adaptive algorithm, e.g. an adaptive feedback cancellation algorithm so as to provide an alternative scheme for feedback estimation in a multi-microphone audio processing system. A feedback part of the system's open loop transfer function is estimated and separated in a transient part and a steady state part, which can be used to control the adaptation rate of the adaptive feedback cancellation algorithm by adjusting the system parameter, e.g. step size parameter, of the algorithm when desired system properties, such as a steady state value or a convergence rate of the feedback, are given/desired. The method can be used for different adaptation algorithms such as LMS, NLMS, RLS, etc. in hearing aids, headsets, handsfree telephone systems, teleconferencing systems, public address systems, etc.

Abstract: Methods, apparatuses, and systems are presented for performing adaptive equalization involving receiving a signal originating from a channel associated with inter-symbol interference, filtering the signal using a filter having a plurality of adjustable tap weights to produce a filtered signal, and adaptively updating each of the plurality of adjustable tap weights to a new value to reduce effects of inter-symbol interference, wherein each of the plurality of adjustable tap weights is adaptively updated to take into account a constraint relating to a measure of error in the filtered signal and a constraint relating to group delay associated with the filter. Each of the plurality of adjustable tap weights may be adaptively updated to drive group delay associated with the filter toward a target group delay.

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 technique for updating filter coefficients of an adaptive filter includes filtering a signal with an adaptive filter, whose filter coefficients are grouped into filter blocks. In this case a number of the filter blocks is less than or equal to a number of the filter coefficients. During each update period, the filter coefficients for less than all of the filter blocks are updated based on a network echo path impulse response.

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 frequency-domain based echo and NEXT canceller is provided. The canceller uses log2 encoding to precondition the error signal representing the echo. An improved gradient constraint is applied on at least a portion of a full weight vector in a least-mean-square algorithm. The least-mean-square algorithm is used to compute filter coefficients. The filter coefficients are multiplied by a frequency-domain data vector using a frequency-domain multiplier to generate frequency-domain output vector.

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: An echo canceller for improved recognition and removal of an echo from a communication device. The echo canceller can dynamically reduce echo using an improved energy estimator and an improved adaptive filter. The improved energy estimator can determine if conversation is in a single talk period or a double talk period based on the combined energy of both the near end background noise and speech. The improved adaptive filter can reduce echo by dynamically changing adaptation speed or step size. In double talk, the adaptive filter(s) can dynamically slow-down or stop adaptation. In single talk, the filter can dynamically increase the speed of adaptation to improve accuracy, or decrease adaptation speed for stability.

Abstract: A method and a system for measuring amplitude and phase difference between two sinusoidal signals, using an adaptive filter. The method generally comprises measuring a sample of an output signal of a system excited by a sample of a reference signal; using an adaptive filter and the sample of the reference signal to determine a and b coefficients that minimize a prediction error on the sample of the output signal, iteratively, and determining the amplitude and/or phase of the output of the system using the a and b coefficients.

Abstract: An adaptive filter configured to use multiple algorithm species that differ in the quality of echo suppression and respective burdens imposed on the computational resources of the host communication device. Depending on the available computational budget, the adaptive filter selects an algorithm species that, while supporting a relatively high quality of echo suppression, involves a relatively low risk of overwhelming the computational resources. The adaptive filter monitors changes in the available computational budget and, if appropriate or necessary, can change the algorithm species to maintain a quality of echo suppression that is optimal for the current computational budget. If a change of the algorithm species is initiated, then at least a portion of internal algorithm data from the previously running algorithm species might be transferred for use in the subsequent algorithm species.

Abstract: Disclosed is an apparatus and method for cancelling a wideband acoustic echo that controls a data flow of data that is transmitted and received using a central processing unit, and calculates data using a calculation processing unit configured to be distinct from the central processing unit.

Abstract: Disclosed is an improved method and apparatus for estimating and applying a step size value for a least mean squares echo canceller. A power estimate of an excitation signal is compared to a reference power level to determine a shift adjustment. The shift adjustment is added to a reference shift amount to determine a shift amount. The product of an excitation signal and an error signal is then calculated and the product is stored in a memory register comprising a plurality of bits. The bits stored in the memory register are shifted either left or right based upon the shift amount. The shift adjustment may be based in part upon the ratio of the excitation signal power estimate and the reference power level.

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: Transmit equalization over high speed digital communication paths may be compensated in a receiver for a probe on that path. In one example, a probe input provides a signal from an electronic communications path, the signal having been processed by a transmit equalizer. A filter circuit processes the signal to compensate for the transmit equalizer, and a decoder decodes the processed signal and produces an output for use by test equipment.

Abstract: A system for removing interference comprising a receive decimation filter that accepts a composite received baseband signal and generates filtered sampled data at a decimation rate, a transmit decimation filter that accepts a digitally converted replica of an interfering signal and generates filtered sampled data at a decimation rate, an integer sample delay control (ISDC) that provides multiple sample delay control for the replica and stores an estimated delay value, an adaptive filter that provides fractional sample delay control for the replica of the interfering signal and optimizes cancellation of the interfering signal, a digital phase-locked loop (DPLL) programmed with a known frequency offset of the interfering signal that tracks a phase and frequency of the replica of the interfering signal, an automatic gain control (AGC) that maintains near full scale operation of adaptive filtering and the DPLL, and a slicer, mixer, and delay unit forming an error estimator.

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: A method and apparatus for cross-talk resistant adaptive noise cancellation.

Abstract: A conferencing endpoint uses acoustic echo cancellation with clock compensation. Receiving far-end audio to be output by a local loudspeaker, the endpoint performs acoustic echo cancellation so that the near-end audio capture by a microphone will lack echo of the far-end audio output from the loudspeaker. The converters for the local microphone and loudspeaker may have different clocks so that their sample rates differ. To assist the echo cancellation, the endpoint uses a clock compensator that cross-correlates an echo estimate of the far-end audio and the near-end audio and adjusts a sample rate conversion factor to be used for the far-end audio analyzed for echo cancellation.

Abstract: Techniques for multi-channel acoustic echo cancellation include adaptive filtering. An adaptive filter can use a lattice predictor of order M coupled to an adaptive LMS/Newton filter of length N, wherein M<N. The lattice predictor can provide decorrelation of the input to the LMS/Newton filter and can provide faster convergence for the LMS/Newton filter. Efficient operation of the LMS/Newton filter can also be provided by using output from the lattice predictor to provide low complexity update of weights for the LMS/Newton filter.

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

Abstract: In a system having an adaptive filter block for receiving a reference signal and adapting to an input signal, and a block for detecting corruption of the input signal by an interference signal and in response limiting adaptation of the adaptive filter block, the improvement comprising delaying and applying the input signal to the adaptive filter block while applying the input signal to the block for detecting corruption without delay.

Abstract: An apparatus generally having a first circuit and a second circuit is disclosed. The first circuit may be configured to generate a first sample by filtering an input vector based on (a) a filter vector and (b) a stochastic vector. Each of a plurality of components in the stochastic vector generally has a respective random value. The first circuit may also be configured to generate a second sample as a difference between a third sample and the first sample. The third sample may be received from a network as an echo. The second circuit may be configured to update a subset of a plurality of taps of the filtering where a corresponding one of the components of the stochastic vector has a first value of the random values.

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: An echo canceler uses an adaptive filter to remove an echo of a far-end input signal from a near-end input signal. Filter coefficients are calculated and updated while the far-end signal is active. While the far-end signal is silent, substitute filter coefficients are similarly calculated to be swapped in as initial values for the filter coefficients when the far-end signal changes from silent to active. The substitute filter coefficients are generated from a simulated far-end signal and a simulated near-end signal obtained by combining corresponding samples in different intervals of the far-end and near-end input signals. To facilitate convergence of the substitute filter coefficients, use of the simulated far-end and near-end signals is disabled when the corresponding samples cancel out.

Abstract: A technique of echo cancellation in a communication system. A method and/or apparatus of echo cancellation that may be suitable for performing echo cancellation under single talk and double talk conditions. A method and/or apparatus of echo cancellation that may significantly reduce a residual echo in a single talk environment (e.g. present in many telecommunications systems) without distorting a near end signal in a double talk environment. A method and/or apparatus of echo cancellation that may reduce a residual echo in single talk and double talk environments by applying a post-processing technique to an ECLMS algorithm.

Abstract: 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 transmit signal second-order inter-modulation (IM2) canceller for a portable handset using a full duplex mode of operation (e.g., WCDMA) is used to controllably reduce IM2 introduced by a transmit signal that appears in a received signal in a receive channel of the portable handset. The transmit signal IM2 canceller includes a delay estimator and a digital signal adjuster. The delay estimator receives a first input from a receive channel and a second input from a transmit channel. The delay estimator generates an estimate of the IM2 that the transmit channel introduces in the receive channel. The digital signal adjuster removes the estimate of the IM2 before forwarding a modified receive channel signal to a baseband subsystem of the portable handset.

Abstract: Embodiments related to echo compensation have been described and depicted.

Abstract: Adaptive filters employing a normalized time domain least mean square algorithm having enhanced convergence rates by virtue of the use of an update gain greater than 2 ? max ? ? or ? ? 2 3 ? ? tr ? [ R ] .

Abstract: A method and apparatus for cross-talk resistant adaptive noise cancellation.

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

Abstract: A method and system for processing subband signals using adaptive filters is provided. The system is implemented on an oversampled WOLA filterbank. Inputs signals are oversampled. The system includes an adaptive filter for each subband, and the functionality of improving the convergence properties of the adaptive filter. For example, the convergence property is improved by whitening the spectra of the oversampled subband signals and/or affine projection algorithm. The system is applicable to echo and/or noise cancellation. Adaptive step size control, adaptation process control using Double-Talk detector may be implemented. The system may further implement a non-adaptive processing for reducing uncorrelated noise and/or cross-talk resistant adaptive noise cancellation.

Abstract: The invention is a method and apparatus for frequency-domain adaptive filtering that has broad applications such as to equalizers, but is particularly suitable for use in acoustic echo cancellation circuits for stereophonic and other multiple channel teleconferencing systems. The method and apparatus utilizes a frequency-domain recursive least squares criterion that minimizes the error signal in the frequency-domain. In order to reduce the complexity of the algorithm, a constraint is removed resulting in an unconstrained frequency-domain recursive least mean squares method and apparatus. A method and apparatus for selecting an optimal adaptation step for the UFLMS is disclosed. The method and apparatus is generalized to the multiple channel case and exploits the cross-power spectra among all of the channels.

Abstract: A multiple channel steered spatialized signal is generated from a signal input modified according to respective spatialization gain functions to generate a plurality of audio channels. An echo cancellation signal is applied to a return path using a combined spatialization and echo path estimate. The estimate is derived from the gain functions applied to the respective channels. When the gain functions applied in the respective channels are changed, for instance to represent a different apparent position of the sound source, a new estimate of the echo paths is generated, based on a previous estimate of the echo path and on the new gain functions.

Abstract: In one aspect of the present invention, a method to reduce noise in a noisy speech signal is disclosed The method comprises: applying at least two versions of the noisy speech signal to a first filter, whereby that first filter outputs a speech reference signal and at least one noise reference signal, applying a filtering operation to each of the at least one noise reference signals, and subtracting from the speech reference signal each of the filtered noise reference signals, wherein the filtering operation is performed with filters having filter coefficients determined by taking into account speech leakage contributions in the at least one noise reference signal.

Abstract: An echo canceller comprises foreground and background filters. The background filter locates and confirms peaks, defines active regions centered about the confirmed peaks and updates coefficients of the foreground filter when the background filter is more effective than the foreground filter A method for improved adaptive echo cancellation comprises configuring a foreground filter, calculating an echo-return loss responsive to the foreground filter, identifying the location of one or more peaks repetitively in a block of data in a background filter, confirming one or more identified peaks over a defined number of blocks before defining an active region equally about the identified peaks in the background filter, filtering the sparse impulse response responsive to the active regions, calculating an echo-return loss responsive to the background filter, and updating the foreground filter when the background filter includes a more effective set of filter coefficients.

Abstract: A method of canceling an echo in a signal of a communication network may include filtering an input signal and a corresponding reference signal of an echo path so as to produce vectors of N subband signals corresponding to the input signal and N subband reference echo signals corresponding to the reference signal. Vectors corresponding to a subset of the N subband signals may be adaptively adjusted to form modified vectors of the subset which approximate the corresponding N subband reference echo signals. Based on the modified vectors, subband errors related to a residual echo for each of the subband signals of the subset may be determined. The subband errors may be synthesized to generate a full-band signal with a synthesized residual echo.

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: In a system having an adaptive filter block for receiving a reference signal and adapting to an input signal, and a block for detecting corruption of the input signal by an interference signal and in response limiting adaptation of the adaptive filter block, the improvement comprising delaying and applying the input signal to the adaptive filter block while applying the input signal to the block for detecting corruption without delay.

Abstract: An echo canceller circuit for use in an echo canceller system is set forth that provides sensitive double-talk detection. The echo canceller circuit comprises a second digital filter having adaptive tap coefficients to simulate an echo response occurring during the call. The adaptive tap coefficients of the second digital filter are updated over the duration of the call using a Least Mean Squares process having an adaptive gain a. A channel condition detector is used to detect channel conditions during the call. The channel condition detector is responsive to detected channel conditions for changing the adaptive gain a during the call. For example, the channel condition detector may detect the presence of a double-talk condition and set the adaptive gain a to zero. Similarly, the channel condition detector may detect the occurrence of a high background noise condition and set the adaptive gain a to a level less than 1 that is dependent on the detected level of the background noise.