Abstract: A receiving apparatus includes: M receivers configured to receive signals based on sound waves propagating in water; M FIR filters configured to perform waveform operation on the signals received by the receivers; a combiner configured to combine output signals of the M FIR filters; and a filter coefficient calculation portion configured to calculate a tap coefficient of the M FIR filters so as to reduce an error of the output signals combined by the combiner. The M FIR filters have a tap length that is shorter than a delay spread that is a possible range between a time of arrival of a direct wave and a time of arrival of a delayed wave of the sound waves.

Abstract: A first physical layer device includes a first transmitter and a first receiver. The first transmitter transmits first data to a second physical layer device over a medium at a first line rate during a first transmit period. The first receiver is configured to not receive data during the first transmit period and an echo reflection period occurring after the first transmit period. The echo reflection period is based on a length of the medium between the first physical layer device and the second physical layer device. The first receiver is configured to, after the echo reflection period, receive second data from the second physical layer device over the medium at a second line rate that is less than the first line rate.

Abstract: A 10BASE-T transmitter includes a Manchester encoder circuit, a waveform shaper circuit, and digital-to-analog converter (DAC) circuit. The Manchester encoder circuit applies Manchester encoding to an input data to generate an encoded data. The waveform shaper circuit converts the encoded data into a plurality of digital codes. The DAC circuit generates a transmit (TX) waveform according to the plurality of digital codes. The waveform shaper circuit controls a portion of the plurality of digital codes for applying pre-compensation of inter-symbol interference (ISI) to the TX waveform.

Abstract: An adaptive filter and an echo cancellation apparatus and method using the same, utilize an adaptive delay diversity filter for coping with the situation in which variation in a time delay between a reference signal and a target signal of an adaptive filter is large. The adaptive delay diversity filter includes multiple sub-filters, each of which generates output through filtering by utilizing a signal, obtained by applying different time delay values to the reference signal, as a reference signal, and selects the best output value among generated output values of the multiple sub-filters, as an output value of the adaptive delay diversity filter. Further, the adaptive delay diversity filter includes a means for reducing a computational load.

Abstract: An electrical balance duplexer (EBD) may be used to isolate a transmitter and receiver that share a common antenna. By using impedance gradients to provide impedances that cause balance-unbalance transformers (balun) of the EBD to cut-off access to the common antenna rather than duplicate the antenna impedance, the EBD is balanced. Such cut-offs may have a lower insertion loss than an EBD that merely duplicates the antenna impedance to separate the differential signals of the receiver/transmitter from the common mode signal.

Abstract: A system for dynamically controlling an echo canceler includes a processor programmed to receive audio data from an audio sensor, to determine a present type of noise in the received audio data, based on the present type of noise, to determine a number of taps to be removed from an echo canceler, and to execute the echo canceler with a reduced number of taps based on the number of taps to be removed.

Abstract: An Ethernet physical-layer circuit corresponding to a first port is connected to a first link partner device through the first port and a first Ethernet cable. The Ethernet physical-layer circuit and other physical-layer circuits all employ an output oscillation signal of a crystal oscillator to respectively generate clock waveforms, and they are configured in a master mode when the crosstalk noise is converged and compensated.

Abstract: An adaptive filter and an echo cancellation apparatus and method using the same, utilize an adaptive delay diversity filter for coping with the situation in which variation in a time delay between a reference signal and a target signal of an adaptive filter is large. The adaptive delay diversity filter includes multiple sub-filters, each of which generates output through filtering by utilizing a signal, obtained by applying different time delay values to the reference signal, as a reference signal, and selects the best output value among generated output values of the multiple sub-filters, as an output value of the adaptive delay diversity filter. Further, the adaptive delay diversity filter includes a means for reducing a computational load.

Abstract: Decision feedback equalization (DFE) tap systems and related apparatuses and methods are disclosed. An apparatus includes output nodes to provide output signals, a complementary metal-oxide-semiconductor (CMOS) DFE tap electrically connected to the output nodes, and a current integrating summer electrically connected to the output nodes. The current integrating summer is to reset the output nodes to a common mode voltage potential.

Abstract: Disclosed herein, among other things, are apparatus and methods for neural network-driven feedback cancellation for hearing assistance devices. Various embodiments include a method of signal processing an input signal in a hearing assistance device to mitigate entrainment, the hearing assistance device including a receiver and a microphone. The method includes performing neural network processing to identify acoustic features in a plurality of audio signals and predict target outputs for the plurality of audio signals, and using the trained neural network to control acoustic feedback cancellation of the input signal.

Abstract: A system for performing echo cancellation includes: a processor configured to: receive a far-end signal; record a microphone signal including: a near-end signal; and an echo signal corresponding to the far-end signal; extract far-end features from the far-end signal; extract microphone features from the microphone signal; compute estimated near-end features by supplying the microphone features and the far-end features to an acoustic echo cancellation module including a recurrent neural network including: an encoder including a plurality of gated recurrent units; and a decoder including a plurality of gated recurrent units; compute an estimated near-end signal from the estimated near-end features; and transmit the estimated near-end signal to the far-end device. The recurrent neural network may include a contextual attention module; and the recurrent neural network may take, as input, a plurality of error features computed based on the far-end features, the microphone features, and acoustic path parameters.

Abstract: A method for detecting an audio input includes: acquiring audio input signals received by at least two input signal channels of an audio input module; for each of the audio input signals, filtering the audio input signal according to a preset audio output signal of an electronic device where the audio input module is located to obtain a target signal; for each of the audio input signals, determining a comparison parameter value according to the target signal and the audio input signal; and determining a performance state of the audio input module according to the comparison parameter values.

Abstract: A method of operating a repeater comprises selecting some of adaptive filter coefficients among a plurality of adaptive filter coefficients of an adaptive filter used for interference cancellation, based on the size of a coefficient, generating a predicted interference signal using the selected adaptive filter coefficients and generating an interference-canceled communication signal from a received signal using the generated predicted interference signal.

Abstract: A first physical layer device includes a first transmitter and a first receiver. The first transmitter transmits first data to a second physical layer device over a medium at a first line rate during a first transmit period. The first receiver is configured to not receive data during the first transmit period and an echo reflection period occurring after the first transmit period. The echo reflection period is based on a length of the medium between the first physical layer device and the second physical layer device. The first receiver is configured to, after the echo reflection period, receive second data from the second physical layer device over the medium at a second line rate that is less than the first line rate.

Abstract: An acoustic echo cancellation device includes: a first echo canceller which, using an input signal obtained from at least two microphones and a reproduced signal outputted to a speaker, produces a first pseudo echo signal which indicates a component of the reproduced signal contained in the input signal; and a second echo canceller which, using at least one input signal outputted from the at least two microphones and the first pseudo echo signal, produces a second pseudo echo signal which indicates a component of the first pseudo echo signal contained in the at least one input signal, and cancels an acoustic echo component of the at least one input signal using the second pseudo echo signal.

Abstract: Echo Cancellation (EC) training for a Full Duplex (FDX) amplifier may be provided. First, a Downstream (DS) signal at a fixed location on a subcarrier frequency may be received. Next, an Upstream (US) echo may be determined from the received DS signal at the fixed location on the subcarrier frequency. Determining the upstream echo may comprise subtracting a known value from the received DS signal at the fixed location on the subcarrier frequency. An Echo Cancelation (EC) coefficient may then be determined. Determining the EC coefficient may comprise dividing the determined US echo by a reference signal comprising an US signal. Next, EC may be performed. Performing EC may comprise subtracting the echo from the DS signal.

Abstract: A noise reduction system includes sensors configured to generate an input signal, an adaptive filter configured to represent a transfer function of a path traversed by the input signal, one or more processing devices, and one or more transducers. The processing devices receive the input signal and generate an updated set of filter coefficients of the adaptive filter by separating the input signal into frequency subbands; determining for each subband, coefficients of a corresponding subband adaptive module; and combining the coefficients of multiple subband adaptive modules. Determining the coefficients of the corresponding subband adaptive module includes selecting a subset of a precomputed set of filter coefficients of the adaptive filter. The processing devices process a portion of the input signal using the updated set of filter coefficients of the adaptive filter to generate an output that destructively interferes with another signal traversing the path represented by the transfer function.

Abstract: A active noise cancellation (ANC) system may include an adaptive filter divergence detector for detecting divergence of the one or more controllable filters as they adapt, based on various temporal or frequency domain amplitude characteristics. Upon detection of a controllable filter divergence, the ANC system may be deactivated, or certain speakers may be muted. Alternatively, the ANC system may modify the diverged controllable filters to restore proper operation of the noise cancelling system. This may include adjusting a leakage value of an adaptive filter controller.

Abstract: Embodiments of the present invention provide a virtual multicarrier design for orthogonal frequency division multiple access communications. Other embodiments may be described and claimed.

Abstract: An audio processing device for use in a network connected audio conferencing system is provided, comprising: a network microphone array comprising two or more microphones (mics) and a beamforming circuit, wherein the network mic array is adapted to acquire acoustic audio signals, convert the same to electric audio signals, perform audio beamforming on the electric audio signals, and output a digital combined beamforming circuit output signal that comprises a first signal part and a second signal part, and wherein the first signal part comprises a first set of digital bits that comprises an active beam index, and wherein the active beam index encodes a selected beam position out of a possible N beam positions, and wherein the second signal part comprises a second set of digital bits that comprises a beamformed audio signal; a receiver adapted to receive the digital combined beamforming circuit output signal and split the same into the first signal part and the second signal part; a plurality of acoustic echo c

Abstract: A system comprises a digital predistortion circuit comprising: a first quantity of delay circuits configured to delay a signal to be predistorted; a second quantity of filter tap circuits, wherein the second quantity is smaller than the first quantity; and a delay-to-filter-taps mapping circuit that is operable to map each output of a first subset of the delay circuits to a corresponding input of the filter tap circuits. The system may comprise circuitry operable to select which of the first quantity of delay circuits is in the first subset. The selection of which of the first quantity of delay circuits is in the first subset may be based on a temperature measurement. The digital predistortion circuit may comprise cross-term generation circuitry operable to generate cross-term signals corresponding to the cross products of multiple, differently-delayed versions of a signal input to the digital predistortion circuit.

Abstract: A method for performing echo cancellation includes: receiving a far-end signal from a far-end device at a near-end device; recording a microphone signal at the near-end device including: a near-end signal; and an echo signal corresponding to the far-end signal; extracting far-end features from the far-end signal; extracting microphone features from the microphone signal; computing estimated near-end features by supplying the microphone features and the far-end features to an acoustic echo cancellation module including: an echo estimator including a first stack of a recurrent neural network configured to compute estimated echo features based on the far-end features; and a near-end estimator including a second stack of the recurrent neural network configured to compute the estimated near-end features based on an output of the first stack and the microphone signal; computing an estimated near-end signal from the estimated near-end features; and transmitting the estimated near-end signal to the far-end device.

Abstract: Communication in a first spectrum and via a first transmission line of first data is according to a time-division duplexing scheme such as G.fast. Communication in a second spectrum and via a second transmission line of second data is according to a frequency-division duplexing scheme such as VDSL2. The first and second spectra both comprise an overlap spectrum. The first transmission line experiences first crosstalk from the second transmission line and the second transmission line experiences second crosstalk from the first transmission line.

Abstract: Communication in a first spectrum and via a first transmission line of first data is according to a time-division duplexing scheme such as G.fast. Communication in a second spectrum and via a second transmission line of second data is according to a frequency-division duplexing scheme such as VDSL2. The first and second spectra both comprise an overlap spectrum. The first transmission line experiences first crosstalk from the second transmission line and the second transmission line experiences second crosstalk from the first transmission line.

Abstract: Acoustic echo cancelling includes receiving a source signal and a sink signal; providing a first error signal representative of an echo-free residual signal based on a first set of coefficients based on the source signal and the sink signal, the first error signal forming an output signal of the controller; providing a second error signal based on a second set of coefficients based on the source signal and the sink signal; detecting a room change if the evaluated first second error signal is greater than a sum or product of the evaluated second first error signal and a first threshold; copying one of sets of reference coefficients stored in a memory to the second acoustic echo canceller; and copying the first set of coefficients from the first acoustic echo canceller as a set of reference coefficients into at least one of the second acoustic echo canceller and the memory.

Abstract: A noise detection method and a noise detection system are provided. The noise detection method includes: obtaining an audio signal; comparing the audio signal with a wave of a noise model to obtain a correlation value; and identifying whether the audio signal is a candidate noise signal based on the correlation value. The method can detect plugging noises effectively.

Abstract: Methods, systems, and media for voice communication are provided. In some embodiments, a system for voice communication is provided, the system including: a first audio sensor that captures an acoustic input; and generates a first audio signal based on the acoustic input, wherein the first audio sensor is positioned between a first surface and a second surface of a textile structure. In some embodiments, the first audio sensor is positioned in a region located between the first surface and the second surface of the textile structure. In some embodiments, the first audio sensor is positioned in a passage located between the first surface and the second surface of the textile structure.

Abstract: Dynamic device speaker tuning for echo control includes detecting audio rendering from a speaker on a device; based at least on detecting the audio rendering, capturing, with a microphone on the device, an echo of the rendered audio; performing a Fourier Transform on the echo and the rendered audio; determining a real-time transfer function for at least one signature band; determining a difference between the real-time transfer function and a reference transfer function; and tuning the speaker for audio rendering, based at least on the difference between the real-time transfer function and the reference transfer function, by adjusting an audio amplifier equalization. For some examples, the signature band represents a wall echo or an alternative mounting option. For some examples, the echo is collected during intervals while the audio rendering is ongoing.

Abstract: Embodiments described herein provide echo cancellation power saving management at a cable transceiver. An echo response signal having a first number of signal components is obtained, via an echo cancellation filter. At a first iteration for calculating a first accumulative echo power, a respective echo tap that corresponds to the first iteration is identified. The first accumulative echo power is calculated for the respective iteration by summing powers of outputs from a last echo tap to the respective echo tap. It is then determined whether the first accumulative echo power, exceeds a pre-determined echo power threshold. If the first accumulative echo power exceeds the pre-determined echo power threshold, a first turn-off indication is sent to the echo cancellation filter to turn off all echo taps including and between the last echo tap to the first echo tap.

Abstract: Audio systems and methods are provided to reduce echo content in an audio signal. The systems and methods receive an audio signal and sound stage rendering parameter(s), and select a set of filter coefficients to filter the audio signal to provide an estimated echo signal. The estimated echo signal is subtracted from a microphone signal to provide an output signal with reduced echo content. The set of filter coefficients are selected based upon the sound stage rendering parameter(s) from among a plurality of stored sets of filter coefficients.

Abstract: A system configured to improve beamforming by using deep neural networks (DNNs). The system can use one trained DNN to focus on a first person speaking an utterance (e.g., target user) and one or more trained DNNs to focus on noise source(s) (e.g., wireless loudspeaker(s), a second person speaking, other localized sources of noise, or the like). The DNNs may generate time-frequency mask data that indicates individual frequency bands that correspond to the particular source detected by the DNN. Using this mask data, a beamformer can generate beamformed audio data that is specific to a source of noise. The system may perform noise cancellation to isolate first beamformed audio data associated with the target user by removing second beamformed audio data associated with noise source(s).

Abstract: In digital video image encoding and decoding, a filter type is selected based on symmetrical properties of the images and coefficient values of an interpolation filter are calculated based on the selected filter type. Coefficient values, filter tap-length and selected filter-type are provided in the encoded video data. Coefficient values are also calculated based on a prediction signal representative of the difference between a video frame and a reference image. The prediction signal is calculated from the reference image based on a predefined base filter and motion estimation performed on the video frame. The predefined base filter has fixed coefficient values. Coefficient values are selected from interpolation of pixel values in a selected image segment in the video frame. Symmetry properties of images can be a vertical symmetry, a horizontal symmetry and a combination thereof, so that only a portion of the filter coefficients are coded.

Abstract: Disclosed is an echo canceller (10) including: a false echo calculator (32) that acquires current and previous filter-coefficient groups used by an adaptive filtering unit (20), and that calculates plural false echo signals by performing a filtering operation using each of the current and previous filter-coefficient groups on a sequence of reception signals (x(n)); a voice-transmission signal buffer that outputs a previous voice-transmission signal (y(n?1)); an evaluation value calculator (34) that calculates plural evaluated values of an echo cancellation quantity by using the previous voice-transmission signal; a filter selector (36) that selects a new filter-coefficient group on the basis of the plural evaluated values of the echo cancellation quantity; a foreground filter (39) that performs a filtering operation using the new filter-coefficient group on the sequence of reception signals (x(n)) to generate an estimated echo component; and a subtractor (25) that subtracts the estimated echo component from a

Abstract: A magnetic resonance imaging (MRI) apparatus for obtaining a magnetic resonance (MR) image using a multi-echo pulse sequence including a plurality of repetition times, including a memory configured to store an MR signal obtained using the multi-echo pulse sequence, and an image processor configured to determine a plurality of echo times included in the plurality of repetition times to provide the multi-echo pulse sequence including the plurality of echo times during the plurality of repetition times, and to obtain the MR image, based on the MR signal, wherein the plurality of repetition times includes a first repetition time adjacent to a second repetition time, wherein the plurality of echo times includes a first echo time of the first repetition time, a second echo time of the first repetition time, a first echo time of the second repetition time, and a second echo time of the second repetition time.

Abstract: A method and apparatus for achieving simultaneous transmit and receive operation with a digital phased array is described. Digital beamforming and cancellation enables adjacent transmitting and receiving sub-arrays to operate simultaneously in the same frequency band.

Abstract: In an audio processing system (300), a filtering section (350, 400): receives subband signals (410, 420, 430) corresponding to audio content of a reference signal (301) in respective frequency subbands; receives subband signals (411, 421, 431) corresponding to audio content of a response signal (304) in the respective subbands; and forms filtered inband references (412, 422, 432) by applying respective filters (413, 423, 433) to the subband signals of the reference signal.

Abstract: A transmission system for transmitting data as part of a communications system, the data comprising a plurality of data symbols or elements, the transmission system being configured to divide the data into at least a first data portion and a second data portion, wherein the first data portion is communicated by transmitting signals in selected carrier channels, wherein the transmission system is configured to encode at least one data symbol or element by selecting a relative order of at least one first carrier channel having a first operational state and at least one second carrier channel having a second operational state.

Abstract: Embodiments of the invention may be used to implement a rate converter that includes: 6 channels in forward (audio) path, each channel having a 24-bit signal path per channel, an End-to-end SNR of 110 dB, all within the 20 Hz to 20 KHz bandwidth. Embodiment may also be used to implement a rate converter having: 2 channels in a reverse path, such as for voice signals, 16-bit signal path per channel, an End-to-end SNR of 93 dB, all within 20 Hz to 20 KHz bandwidth. The rate converter may include sample rates such as 8, 11.025, 12, 16, 22.05, 24, 32 44.1, 48, and 96 KHz. Further, rate converters according to embodiments may include a gated clock in low-power mode to conserve power.

Abstract: A portable device performs echo cancellation and echo suppression. An audio echo signal and an audio desired signal are obtained from an acoustic echo correction stage of the portable device. The echo and desired signals are converted to the frequency domain. Frequency bin results of the respective frequency domain converted echo and desired signals are grouped into echo and desired sub-bands. A sub-band suppressor gain is estimated based on the estimated sub-band energy for the echo and desired sub-bands. The frequency domain converted echo signal is modulated based at least in part on the estimated sub-band suppressor gain to compensate for residual echo. The compensated frequency domain converted echo signal is time domain converted into an audio output signal. The audio output signal is processed by a selected one of a voice recognition engine and a communication module transmitter.

Abstract: A controller for an echo suppressor configured to suppress a residual echo of a far-end signal included in a primary error signal, the controller adapted for operation with a primary adaptive filter configured to form a primary echo estimate of the far-end signal included in a microphone signal and an echo canceller configured to cancel that primary echo estimate from the microphone signal so as to form the primary error signal, the controller comprising: a secondary adaptive filter configured to form a secondary echo estimate of the far-end signal comprised in the microphone signal; and control logic operable in at least two modes selected in dependence on a convergence state of the primary adaptive filter, the control logic being configured to control activation of the echo suppressor in dependence one or more transient or steady state decision parameters.

Abstract: In accordance with embodiments of the present disclosure, a system may include an audio input for receiving an audio input signal for reproduction at a speaker, a sensor input for receiving at least one sensor signal indicative of a physical quantity indicative of a variable acoustic volume to which an acoustic output of a speaker generates sound, wherein the acoustic volume is partially enclosed by structural members of an information handling system comprising the speaker, an output for generating an audio output signal to the speaker, and a processor configured to apply an adaptive audio response to the audio input signal to generate the audio output signal, wherein the adaptive audio response is a function of the physical quantity.

Abstract: Technical solutions are described for acoustic echo cancellation. An example method includes computing, by a beamformer, a plurality of updated beamformer filter coefficients, the beamformer filter coefficients computed adaptively to determine a speech signal from a plurality of input audio signals. Further, the method includes computing, by a transfer function estimator, a relative transfer function based on the updated beamformer filter coefficients and a plurality of present acoustic echo canceller coefficients. Further, the method includes adjusting a plurality of acoustic echo canceller coefficients using the relative transfer function, and generating an output speech signal, by the acoustic echo canceller, by cancelling echo components from the speech signal using the adjusted filter coefficients. Further, the method includes sending, by the acoustic echo canceller, the output speech signal to a far end speech device.

Abstract: Method for performing speech enhancement using a Deep Neural Network (DNN)-based signal starts with training DNN offline by exciting a microphone using target training signal that includes signal approximation of clean speech. Loudspeaker is driven with a reference signal and outputs loudspeaker signal. Microphone then generates microphone signal based on at least one of: near-end speaker signal, ambient noise signal, or loudspeaker signal. Acoustic-echo-canceller (AEC) generates AEC echo-cancelled signal based on reference signal and microphone signal. Loudspeaker signal estimator generates estimated loudspeaker signal based on microphone signal and AEC echo-cancelled signal. DNN receives microphone signal, reference signal, AEC echo-cancelled signal, and estimated loudspeaker signal and generates a speech reference signal that includes signal statistics for residual echo or for noise.

Abstract: An echo canceller device is provided with a first filter unit that generates a first output signal y1(k) by filtering a first input signal x1(k) with a filter coefficient h(k), an adaptation unit that modifies the filter coefficient h(k) on the basis of an adaptation algorithm so that a first error signal e1(k) becomes small, a first delaying unit that delays a first objective signal d1(k) by a certain time period ? and outputs the signal as a second objective signal d2(k), a second delaying unit that delays the first input signal x1(k) by the certain time period ? and outputs the signal as a second input signal x2(k), and a second filter unit that generates a second output signal y2(k) by filtering the second input signal x2(k) by the filter coefficient h(k) modified by the adaptation unit.

Abstract: A full-duplex transceiver is provided with componentry and methods for cancellation of nonlinear self-interference signals. The transceiver is capable of receiving an incoming radio-frequency signal that includes both a desired radio-frequency signal component and a self-interference component caused by the transceiver's own radio-frequency transmission. The transceiver demodulates the incoming radio-frequency signal to generate a first demodulated signal. The transceiver combines an analog corrective signal with the first demodulated signal to generate a second demodulated signal with reduced self-interference. The transceiver processes the first and second demodulated signals to determine a desired incoming baseband signal and to determine nonlinear components of the self-interference signal, such as nonlinearities introduced by the transceiver's power amplifier.

Abstract: Embodiments of the present invention provide a virtual multicarrier design for orthogonal frequency division multiple access communications. Other embodiments may be described and claimed.

Abstract: This disclosure provides systems, methods, and apparatus for mitigating the effects of interference signals on optical signals received at a direct-detection optical receivers. The optical receivers are capable of attenuating interference noise signals resulting from the interference between a transmitted optical signal transmitted from a transmitter to the optical receiver and one or more additional signals received at the optical receiver. The interference can be due to multi-path interference or due to in-band interference. The receivers include a tunable filter for filtering the received optical signal to remove the interference. A frequency offset module processes the received optical signal to determine a frequency offset indicative of the difference between the carrier frequencies of a modulated optical signal and an interference optical signal.

Abstract: An echo cancellation system performs audio beamforming to separate audio input into multiple directions (e.g., target signals) and generates multiple audio outputs using two acoustic echo cancellation (AEC) circuits. A first AEC removes a playback reference signal (generated from a signal sent a loudspeaker) to isolate speech included in the target signals. A second AEC removes an adaptive reference signal (generated from microphone inputs corresponding to audio received from the loudspeaker) to isolate speech included in the target signals. A beam selector receives the multiple audio outputs and selects the first AEC or the second AEC based on a linearity of the system. When linear (e.g., no distortion or variable delay between microphone input and playback signal), the beam selector selects an output from the first AEC based on signal to noise (SNR) ratios. When nonlinear, the beam selector selects an output from the second AEC.

Abstract: One or more delay estimators are used to detect failure in a signal processing component, such as an acoustic echo canceller. A first delay estimator is used to generate i) an estimated post-processing delay between when a known audio signal was conveyed to a loudspeaker, and when a portion of the processed audio signal that includes the known audio signal was output from the signal processing component, and ii) a confidence level for the estimated post-processing delay. A failure of the signal processing component may be detected in response to the estimated post-processing delay exceeding a threshold. A second delay estimator may also be used to generate an estimated pre-processing delay and a confidence level for the estimated pre-processing delay for comparison to the estimated post-processing delay and confidence level for the estimated post-processing delay in order to provide further failure detection accuracy and specificity.

Abstract: A variable update step size is determined in proportion to a magnitude ratio or magnitude difference between a first residual signal and a second residual signal. The first residual signal is obtained by using adaptive filter coefficient sequence, where the adaptive filter coefficient sequence has been obtained in previous operations of the adaptive equalizer. The second residual signal is obtained by using a prior update adaptive filter coefficient sequence, where the prior update adaptive filter coefficient sequence is obtained by performing a coefficient update with an arbitrary prior update step size on the adaptive filter coefficient sequence having been obtained in previous operations of the adaptive equalizer.