Abstract: Example techniques involve noise-robust acoustic echo cancellation. An example implementation may involve causing one or more speakers of the playback device to play back audio content and while the audio content is playing back, capturing, via the one or more microphones, audio within an acoustic environment that includes the audio playback. The example implementation may involve determining measured and reference signals in the STFT domain. During each nth iteration of an acoustic echo canceller (AEC): the implementation may involve determining a frame of an output signal by generating a frame of a model signal by passing a frame of the reference signal through an instance of an adaptive filter and then redacting the nth frame of the model signal from an nth frame of the measured signal. The implementation may further involve determining an instance of the adaptive filter for a next iteration of the AEC.

Abstract: The present invention discloses a signal transceiving apparatus having echo-canceling mechanism. A mixer circuit includes a Wheatstone bridge and a transformer winding circuit. The Wheatstone bridge includes another transformer winding circuit and a variable load and includes a first input terminal, a first output terminal, a second input terminal and a second output terminal located at each two neighboring arms in an order. A transmission circuit is coupled to the first input terminal and the second input terminal to perform signal transmission through the mixer circuit. A receiving circuit is coupled to the first output terminal and the second output terminal to perform signal receiving through the mixer circuit. A control circuit adjusts the impedance of the variable load when a residual echo noise amount does not satisfy a minimum echo noise amount condition, and stops to adjust the impedance when the residual echo noise amount satisfies the condition.

Abstract: Example techniques involve noise-robust acoustic echo cancellation. An example implementation may involve causing one or more speakers of the playback device to play back audio content and while the audio content is playing back, capturing, via the one or more microphones, audio within an acoustic environment that includes the audio playback. The example implementation may involve determining measured and reference signals in the STFT domain. During each nth iteration of an acoustic echo canceller (AEC): the implementation may involve determining a frame of an output signal by generating a frame of a model signal by passing a frame of the reference signal through an instance of an adaptive filter and then redacting the nth frame of the model signal from an nth frame of the measured signal. The implementation may further involve determining an instance of the adaptive filter for a next iteration of the AEC.

Abstract: An echo path monitoring system for controlling an adaptive filter configured to estimate an echo of a far-end signal comprised in a microphone signal, the system comprising a comparison generator configured to compare the microphone signal with the estimated echo to obtain a first comparison and compare an error signal, which represents a difference between the microphone signal and the estimated echo, with the estimated echo to obtain a second comparison, and a controller configured to combine the first and second comparisons to form a parameter indicative of a state of the microphone signal and, in dependence on said parameter, control an operating mode of the adaptive filter.

Abstract: An audio processor for a video conference system receives an audio signal from content to be shared over a video conference and an audio signal from a network. The audio signal from the shared content and the audio signal from the network are mixed together for output to a speaker. The audio processor may also receive a local audio signal from a microphone. The local audio signal is mixed with the audio signal of the shared to content to generate an outbound signal.

Abstract: Techniques are provided for deriving speed estimates by performing a convolution of the output of the Fast Fourier Transform (FFT) stage from different time windows in an orthogonal frequency-division multiplexing (OFDM) modulation scheme. Such techniques enable speed estimation to be performed in the frequency domain, requiring little or no modification to existing OFDM systems.

Abstract: Disclosed in the invention is a method and system for sampling rate mismatch correction of transmitting and receiving terminals, which can obtain a high-precision sampling rate mismatch in real time, carry out sampling rate correction on transmitting and receiving terminal signals, and send the transmitting terminal signal and the receiving terminal signal that have the same sampling rate after corrected to an echo cancellation system to carry out echo cancellation. The present invention can improve the quality of echo cancellation, simplify the computation and reduce the cost.

Abstract: A acoustic echo canceller (AEC) system may be configured to perform echo cancellation in the frequency domain. Features are disclosed for determining an estimated echo in the frequency domain using adaptive filters. An adaptive filter corresponding to a frequency bin can comprise a plurality of filter taps. Additional features are disclosed for updating the adaptive filter. In addition, a frequency-bin dependent step size controller may be used to control a step size used in updating the adaptive filters. Features are disclosed for determining the frequency-bin dependent step size.

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 acoustic echo suppression unit according to an embodiment of the present invention includes and input interface for extracting a downmix signal from an input signal, the input signal including the downmix signal and parametric side information, wherein the downmix and the parametric side information together represent a multichannel signal, a calculator for calculating filter coefficients for an adaptive filter, wherein the calculator is adapted to determine the filter coefficients based on the downmix signal and a microphone signal or a signal derived from the microphone signal, and an adaptive filter adapted to filter the microphone signal or the signal derived from the microphone signal based on the filter coefficients to suppress an echo caused by the multichannel signal in the microphone signal.

Abstract: An apparatus includes an echo canceller having an audio signal input and an audio signal output and dynamic pre-conditioning logic. The dynamic pre-conditioning logic is operatively coupled to the echo canceller audio signal output as a feedback signal and has a dynamic pre-conditioning logic output operatively coupled to the echo canceller audio signal input. The dynamic pre-conditioning logic is also operative to receive an audio signal input from at least one microphone. The dynamic pre-conditioning logic is operative to analyze the feedback signal to obtain at least one characteristic, and pre-condition the audio signal input, based on the at least one characteristic of the feedback signal, and provide a pre-conditioned audio signal at the echo canceller audio signal input. The echo canceller audio signal output is then provided to a noise suppressor for the send path of a full duplex communication channel.

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

Abstract: A system and method are presented for acoustic echo cancellation. The echo canceller performs reduction of acoustic and hybrid echoes which may arise in a situation such as a long-distance conference call with multiple speakers in varying environments, for example. Echo cancellation, in at least one embodiment, may be based on similarity measurement, statistical determination of echo cancellation parameters from historical values, frequency domain operation, double talk detection, packet loss detection, signal detection, and noise subtraction.

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: An echo cancelling algorithm in a communication device initializes a step size value used in an adaptive echo filter based on a background noise signal power level relative to a power level of a received signal and a power level of an echo estimate relative to an output of an echo canceller. The algorithm then adjusts the step size value. One aspect adjusts the step size based on the detection of large fast fourier transform values at one, or more, disturbing-signal frequencies. Another aspect estimates residual echo energy to adjust an estimated echo energy, which then is used to set a double talk flag if a transmit signal has much more power than the estimated echo signal. Another aspect compares transmit signal power to a decimated version of the transmit signal power and sets the double talk flag if the former exceeds the latter by a predetermined amount.

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

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

Abstract: 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 preferred embodiment of an apparatus for computing filter coefficients for an adaptive filter for filtering a microphone signal so as to suppress an echo due to a loudspeaker signal includes an extractor for extracting a stationary component signal or a non-stationary component signal from the loudspeaker signal or from a signal derived from the loudspeaker signal, and a computer for computing the filter coefficients for the adaptive filter on the basis of the extracted stationary component signal or the extracted non-stationary component signal.

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: 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 clipping detection device calculates an amplitude distribution of an input signal for each predetermined period, calculates a deflection degree of the distribution on the basis of the calculated amplitude distribution, and then detects clipping of a communication signal on the basis of the calculated deflection degree of the distribution.

Abstract: A method and corresponding apparatus for coded-domain acoustic echo control is presented. An echo control problem is considered as that of perceptually matching an echo signal to a reference signal. A perceptual similarity function that is based on the coded spectral parameters produced by the speech codec is defined. Since codecs introduce a significant degree of non-linearity into the echo signal, the similarity function is designed to be robust against such effects. The similarity function is incorporated into a coded-domain echo control system that also includes spectrally-matched noise injection for replacing echo frames with comfort noise. Using actual echoes recorded over a commercial mobile network, it is shown herein that the similarity function is robust against both codec non-linearities and additive noise. Experimental results further show that the echo-control is effective at suppressing echoes compared to a Normalized Least Mean Squared (NLMS)-based echo cancellation system.

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: In an echo suppression apparatus, an adaptive filter estimates an echo path of a near end, and generates a pseudo echo signal of a reception signal received from a far end. A subtractor subtracts the pseudo echo signal from a near-end signal including an echo signal of the reception signal, a near-end sound and a background noise, thereby generating an echo-canceled signal. A background noise estimation unit estimates a spectrum of the background noise. A non-linear processing unit performs spectrum subtraction of the estimated spectrum of the background noise from the spectrum of the echo-canceled signal, and controls a gain of the spectrum of the echo-canceled signal in response to the result of the spectrum subtraction, thereby obtaining a spectrum of a transmission signal transmitted to the far end. A threshold calculation unit calculates a threshold value used to determine presence or absence of the residual echo in the echo-canceled signal.

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 double talk detector for controlling the echo path estimation in a telecommunication system by indicating when a received coded speech signal is dominated by a non-echo signal; i.e., that so-called double talk exists. This is determined by extracting LSPs from a coded speech frame of the received coded speech signal when the signal power exceeds a first threshold value, converting each of said extracted LSPs into LSFs, and calculating the distance between each two adjacent LSFs. For each distance that is smaller than a second threshold, a spectral peak is located between the two LSFs, and it is determined whether said spectral peak is an echo or not. When a predetermined number of non-echo spectral peaks are located in the received speech signal, double talk will be indicated, and the echo path estimation may be disabled.

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: Embodiments of an 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 a training signal 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. The training signal may be generated from a spectrum estimate and a training sequence. An adaptive filter may generate an echo-cancellation signal using filter coefficients generated from the training sequence and return-path signals. The echo-cancellation signal may remove echo signals from the return-path signals.

Abstract: Harmonic distortion residual echo suppression (HDRES) technique embodiments are presented which act to suppress the residual echo remaining after a near-end microphone signal has undergone AEC, including harmonic distortion in the signal that was caused by the speaker audio signal playback. In general, an AEC module is employed which suppresses some parts of the speaker audio signal found in a near-end microphone signal and generates an AEC output signal. A HDRES module then inputs the AEC output signal and the speaker audio signal, and suppresses at least a portion of a residual part of the speaker audio signal that was left unsuppressed by the AEC module. This includes at least a portion of the harmonic distortion exhibited in the AEC output signal.

Abstract: In a speech communication device capable of bidirectional communication, a clock deviation between the speaker side (the receiving side) and the microphone side (the transmitting side) is detected, and based on the above deviation, either a first frequency signal (speaker signal (reference signal)) or a second frequency signal (microphone signal) is frequency-shifted in the frequency domain, and therefore, an influence caused by the clock deviation can simply be corrected without an increased processing amount, and thus, an echo can be suppressed effectively.

Abstract: An acoustic echo cancellation device for canceling an echo in a microphone signal in response to first and second input signals includes a first combination unit for combining the first and second input signals into an aggregate input signal. The device further includes an adaptive filter unit for filtering the aggregate input signal so as to produce an aggregate echo cancellation signal. A second combination unit combines the aggregate echo cancellation signal with the microphone signal so as to produce a residual signal, and an additional filter unit filters either the first or second input signal so as to produce a first or a second partial echo cancellation signal. A post-processor unit suppresses remaining echo components in the residual signal. The post-processor unit uses at least one partial echo cancellation signal to suppress echo components corresponding with the first input signal to a greater extent than echo components corresponding with the second input signal.

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 method and apparatus for removing an echo signal in a signal transmission/reception apparatus of a communication system is provided. A signal transmission/reception apparatus estimates an input channel response using a training sequence, generates a first signal by removing the input channel response from a first reception signal, detects an echo channel impulse response using the first signal, detects an echo signal removing coefficient using the echo channel impulse response, generates a second signal in which an echo signal is removed by applying the echo signal removing coefficient to a second reception signal, and removes the second signal from a third signal, wherein the first signal is received prior to receiving the second signal and the second signal is received prior to receiving the third signal.

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: Hybrid echo canceller controllers are described herein. By way of example, a system for controlling an echo canceller can include a signal indicator and an echo canceller controller. The signal indicator can be configured to indicate periods of near-end signal and to indicate periods of echo only with echo-path change in the corrupted signal based at least in part on cross-correlation between two signals associated with the echo canceller. The echo canceller controller can be configured to control the echo canceller according to indications from the signal indicator.

Abstract: There is disclosed a multi-carrier transceiver system for use in echo cancellation. The transceiver system is arranged to generate from input data a multicarrier transmit signal comprising a plurality of multi-carrier data blocks. The transceiver system also has a multi-carrier signal receiver responsive to an input multi-carrier receive signal received from a remote signal transmitter. An echo canceller is arranged to generate from a pair of multi-carrier data blocks (u) a set of frequency-domain echo parameters for use in generating an echo signal, to generate the echo signal using the frequency-domain echo parameters, and to input the echo signal to the multi-carrier signal receiver for use in generating an echo-cancelled receive signal from the input receive signal and the echo signal.

Abstract: A hybrid circuit for a full-duplex transceiver includes a two-to-four wire ratio converter (50, Z1, Z2) for forwarding a transmitted signal TX? on a transmission line (41) in response to an input signal TX. The ratio converter also produces a signal RXA in response to a received signal RX arriving on the transmission line, wherein signal RXA includes an echo of input signal TX. A passive filter (H1) and a first active filter (H2) each filter input signal TX to generate signals TXA and TXB summed with the RXA signal to form a signal RXB at the input of an amplifier (52, Z3, Z4) producing an output signal RXC. A second active filter (H3) filters input signal TX to generate a signal TXC summed with the RXC signal to produce the output signal RX. The impedance of the passive filter is designed so that when the transmission line has a target impedance, such as the impedance of an ideal twisted pair, the TXA output signal of the passive filter adequately offsets the TX signal echo in RXA.

Abstract: A method and corresponding apparatus for coded-domain acoustic echo control is presented. An echo control problem is considered as that of perceptually matching an echo signal to a reference signal. A perceptual similarity function that is based on the coded spectral parameters produced by the speech codec is defined. Since codecs introduce a significant degree of non-linearity into the echo signal, the similarity function is designed to be robust against such effects. The similarity function is incorporated into a coded-domain echo control system that also includes spectrally-matched noise injection for replacing echo frames with comfort noise. Using actual echoes recorded over a commercial mobile network, it is shown herein that the similarity function is robust against both codec non-linearities and additive noise. Experimental results further show that the echo-control is effective at suppressing echoes compared to a Normalized Least Mean Squared (NLMS)-based echo cancellation system.

Abstract: Disclosed is a non-linear echo canceller and method for cancelling echo during full duplex communication in a hands free communication system. An input signal from a far-end talker and an input signal from the output from an echo canceller are received. K spectral subbands are created for each input signal. The spectral echo residual power at each subband is estimated and compared to a clean signal power to calculate a signal to echo ratio. Gains are calculated based on each calculated ratio and non-linear echo is cancelled based on the calculated gains.

Abstract: The present invention relates to detection of echo in telecommunications networks. The invention provides a method of echo detection comprising the steps of: generating a series of inbound vectors for an inbound signal; generating a series of outbound vectors for an outbound signal; repeating a predetermined number of comparison steps comprising the sub-steps of selecting an outbound vector from the outbound vectors; selecting an inbound vector from the inbound vectors; comparing said outbound vector with said inbound vector and with successive inbound vectors to generate a plurality of similarity metrics; and determining a relative position of the compared outbound vector having maximum correlation with said inbound vector; and counting the number of times each relative position is determined to be the position of maximum correlation.

Abstract: A method to design low complexity and low power echo and NEXT cancellers based on wordlength reduction technique is presented. A circuit architecture to implement echo and cancellers is also presented. The low complexity and low power design relies on the fact that a TH precoder can be viewed as an IIR filter with an input equal to the sum of the original input to the TH precoder and a compensation signal. The proposed design also relies on the fact that sum of the original input to the TH precoder and the compensation signal has finite levels, which can be represented in less bits than the original input of the echo and NEXT cancellers. An improved design by exploiting the statistics of the compensation signal is also proposed to further bring down the complexity and power consumption of these cancellers.

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 cancellation technique that can process multi-input microphone signals with only a small increase in the overall CPU consumption compared to implementing the algorithm for a single channel microphone signal. Furthermore, the invention provides an architecture that provides for echo cancellation for multiple applications in parallel with only a small increase in CPU consumption compared to a single instance of echo cancellation with a single microphone input and multi-output channel playback.

Abstract: A regression-based residual echo suppression (RES) system and process for suppressing the portion of the microphone signal corresponding to a playback of a speaker audio signal that was not suppressed by an acoustic echo canceller (AEC). In general, a prescribed regression technique is used between a prescribed spectral attribute of multiple past and present, fixed-length, periods (e.g., frames) of the speaker signal and the same spectral attribute of a current period (e.g., frame) of the echo residual in the output of the AEC. This automatically takes into consideration the correlation between the time periods of the speaker signal. The parameters of the regression can be easily tracked using adaptive methods. Multiple applications of RES can be used to produce better results and this system and process can be applied to stereo-RES as well.

Abstract: The encoder transforms the audio signals (x(n),y(n)) from the time domain to audio signal (X(k),Y(k)) in the frequency domain, and determines the cross-correlation function (Ri, Pi) in the frequency domain. A complex coherence value (Qi) is calculated by summing the (complex) cross-correlation function values (Ri, Pi) in the frequency domain. The inter-channel phase difference (IPDi) is estimated by the argument of the complex coherence value (Qi), and the inter-channel coherence (ICi) is estimated by the absolute value of the complex coherence value (Qi).