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: The present invention relates to a device, such as a communication device, comprising an adaptive foreground filter configured to calculate a first echo estimation signal based on a first input signal, and an adaptive background filter being more rapidly adapting than the foreground filter and configured to calculate a second echo estimation signal based on said first input signal. Embodiments of the device further comprise damping control means for controlling damping of an echo-cancelled output signal. The device in various embodiments includes that the damping control means is configured to calculate a maximum echo estimation signal using both the first and the second echo estimation signals, and control the damping of the echo-cancelled output signal based on said maximum echo estimation signal and/or a signal derived from said maximum echo estimation signal.

Abstract: Techniques to reduce distortion in acoustic signals in mobile computing devices are described. For example, a mobile computing device may comprise a speaker operative to receive a first signal and output a second signal. The mobile computing device may further comprise a first microphone operative to receive the second signal and a second microphone operative to receive a third signal. An echo canceller may be coupled to the first microphone and the second microphone and may be operative to compare the second signal and the third signal and reduce distortion in the third signal based on the comparison. Other embodiments are described and claimed.

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: Methods and systems for a dual echo canceller (EC) are disclosed and may include cancelling echo in utilizing a dual echo canceller, wherein said dual echo canceller includes an active echo canceller and an adaptive echo canceller. Filter coefficients may be copied from the adaptive echo canceller to the active echo canceller for the cancellation, based on whether said adaptive echo canceller has converged. The coefficients may be copied utilizing copy logic, which may comprise divergence detection and/or echo path change detection. The coefficients may be reset to default settings utilizing the copy logic. The coefficients may be calculated utilizing normalized block least mean squares (NBLMS), and may be calculated when the NBLMS is enabled by update logic. The coefficients may be calculated utilizing linear predictive coefficient (LPC) filtered uplink and downlink signals.

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: In an example, time and frequency domain speech enhancement is implemented on a platform having a programmable device, such a PC or a smartphone running an OS. Echo cancellation is done first in time domain to cancel a dominant portion of the echo. Residual echo is cancelled jointly with noise reduction during a subsequent frequency domain stage. The time domain block uses a dual band, shorter length Adaptive Filter for faster convergence. Non-linear residual echo is cancelled based on an echo estimate and an error signal from the adaptive filters. A controller locates regions that had residual echo suppressed and which do not have speech and injects comfort noise. The controller can be full-duplex and operate non-linearly. An AGC selectively amplifies the frequency bins, based on the Gain function used by the residual echo and noise canceller.

Abstract: In one or more embodiments, an echo reduction system can include multiple adaptive filters operable to provide estimated echo replicas based on received signal information coupled to respective multiple adders that can combine a send signal and outputs of the respective adaptive filters and can provide respective output combinations to a filter selector. The filter selector can select from the outputs of the adders and provide a selected output as output for the echo reduction system. In one or more embodiments, the filter selector can control signal processing of the filters by providing control signal to the adaptive filters, where the control signals can indicate to ones of the adaptive filters to pause or continue signal processing. In one or more embodiments, the echo reduction system can include at least one delay unit that can delay receive signal information to at least one of the adaptive filters.

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: An echo canceling system has an adaptive filter connected to an echo source, for canceling echo from an echo path. A fixed echo conditioning filter is connected in series with the adaptive filter, intermediate the echo source and the adaptive filter. The fixed echo conditioning filter cancels the constant part of the echo path using a measured impulse response of the echo path.

Abstract: A signal generator produces a victim signal having crosstalk emulation by filtering and combining a victim signal waveform record file and an aggressor signal waveform record file generated using parameters selected by a user. A signal channel or a cascaded signal channel is characterized using one or more S-parameter arrays. The S-parameter array or arrays represent a mixed-mode multiple-port device under test. Coefficients of a NEXT filter, a FEXT filter and a forward transmission filter are derived from selected S-parameters of the S-parameter array. The aggressor signal is filtered individually by the NEXT and FEXT filters. The victim signal is summed with the filtered aggressor signal from the NEXT filter with the resulting summed signal being filtered by the forward transmission filter. The filtered signal from the forward transmission filter is summed with the filtered aggressor signal from the FEXT filter to generate a victim signal having crosstalk emulation.

Abstract: An echo canceler 10 generates an echo elimination signal by filtering through adaptive filters 101 and 102 reference signals input from sound sources causing echoes. It includes a sound source number detecting unit 103 for detecting the number of the sound sources causing echoes from the reference signals, and a control unit 105 for making the number of taps of the adaptive filters 101 and 102 variable in accordance with the number of the sound sources detected by the sound source number detecting unit 103.

Abstract: An echo canceler 10 generates an echo elimination signal by filtering through adaptive filters 101 and 102 reference signals input from sound sources causing echoes. It includes a sound source number detecting unit 103 for detecting the number of the sound sources causing echoes from the reference signals, and a control unit 105 for making the number of taps of the adaptive filters 101 and 102 variable in accordance with the number of the sound sources detected by the sound source number detecting unit 103.

Abstract: A digital subscriber line access multiplexer (DSLAM) configured to couple to a first subscriber line and a second subscriber line, wherein the DSLAM comprises at least one transmitter configured to transmit a first signal onto the first subscriber line in a first time slot, transmit a second signal onto the second subscriber line in the first time slot, wherein the first and second signals are linearly related and are intended for a first customer premise equipment (CPE), transmit a third signal onto the second subscriber line in a second time slot, and transmit a fourth signal onto the first subscriber line in the second time slot, wherein the third and fourth signals are linearly related and are intended for a second CPE.

Abstract: In a telephone device, a near-end signal processor delays each input near-end signal delivered from the respective microphones by a delay amount, and synthesizes the input near-end signals into a synthesized near-end signal. In an echo canceller, mounted on board the telephone device, a far-end signal is delayed by the same delay amounts as used in the near-end signal processor to form plural delayed far-end signals, which are synthesized into a synthesized far-end signal, with which the echo component in the synthesized near-end signal will be suppressed.

Abstract: If K=4N, wherein K is a number of subbands and N is a decimation ratio, an analysis filter process unit 21 is used to perform an analysis filter process for every N samples of real-valued input data and output K samples of real-valued data. A GDFT/SSB modulation batch process unit (81) performs a GDFT process and an SSB modulation process on the outputted data by using a real-valued matrix calculation using a (K/2)×K real-valued matrix to output 1 sample×(K/2) channel of real-valued subbands data. Moreover, an SSB demodulation/inverse GDFT batch process unit (82) performs an SSB demodulation process and an inverse GDFT process on 1 sample×(K/2) channel of real-valued subbands data by a real-valued matrix calculation using a K×(K/2) real-valued matrix and outputs K samples of real-valued data. The outputted data is subjected to a synthesis filter process by a synthesis filter process unit (24) to output N samples of real-valued data.

Abstract: An echo removing apparatus that can remove an echo appropriately if the produced sound volume becomes large although the configuration is simple is provided. A control section 17 switches the operations of an echo suppressor 14 and an echo canceller 15 in response to the volume value input from a volume operation section 18. If the volume value is large, the echo suppressor 14 is operated and an echo is suppressed reliably. If the volume value is small, the echo canceller 15 is operated and double talk is enabled.

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: Methods and systems for updating the adaptive filter of an echo canceller. A method of updating the adaptive filter of an echo canceller in which an estimated echo is resolved from a received signal and then subtracted from an incoming echo-contaminated signal so as to produce a filtered output signal, includes: obtaining a corrected impulse response of an echo reconstruction filter (ERF); calculating specified decision measures usable to decide whether to prospectively apply the corrected ERF impulse response or a current ERF impulse response; determining whether application of the corrected ERF impulse response would result in improved echo cancellation; and updating the ERF to apply the corrected impulse response, when it is determined that the updating would result in a lower residual echo.

Abstract: A speakerphone having transmit path with a microphone and a first amplifier having a gain determined by a controller and an input coupled to the microphone, a receive path with a second amplifier having a gain determined by the controller and a speaker coupled to an output of the second amplifier, and operable in a half-duplex mode using an adaptive echo canceller to at least partially remove from the transmit path acoustically-coupled signals from the receive path. The controller increases the gain of the first amplifier and decreases the gain of the second amplifier when a level of signals in the transmit path exceed a level of signals in the receive path, and decreases the gain of the first amplifier and increases the gain of the second amplifier when the level of signals in the receive path exceed the level of signals in the transmit path.

Abstract: A multi-party spatial audio conferencing system is configured to receive far end signals from remote participants. The system comprises a speaker array that outputs spatialized sound signals and one or more microphones that capture and relay a sound signal comprising an echo of the spatialized sound signal to a multichannel acoustic echo cancellation (MC-AEC) unit having a plurality of echo cancellers. Respective echo cancellers perform cancellation of an echo associated with a far end signal from one of the multiple participants according to an algorithm based upon echo cancellation coefficients. The echo cancellation coefficients are determined from the input channel signals, the spatialization parameters associated with each input channel, and the audio signals captured by the microphones. This allows respective echo cancellation filters to be updated simultaneously even though the corresponding remote participant is not talking.

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

Abstract: Example methods, apparatus and articles of manufacture to detect echo during teleconferences are disclosed. A disclosed example method includes collecting first signal data for a first teleconference leg of a teleconference, collecting second signal data for a second teleconference leg of the teleconference, computing an echo detection metric from the first and second signal data, and comparing the echo detection metric to a threshold to determine whether an echo is likely present on the first teleconference leg of the teleconference.

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

Abstract: Methods and apparatus of managing a communication system, wherein a decision regarding a level of activity at a first end is made based at least in part on the level of activity at the second end. In one embodiment, the energy level of a first-end audio signal is measured. The first end is declared voice-active if the first-end energy level is greater than or equal to a first threshold value. The first end is declared voice-inactive if the first-end energy level is less than the first threshold value. To determine the value of the first threshold value, the energy level of a second-end audio signal is measured. If the second-end energy level is greater than or equal to a second threshold value, the second end is declared voice-active, in which case the first threshold is maintained at a relatively high level. If the second-end energy level is less than the second threshold value, the second end is declared voice-inactive, in which case the first threshold is maintained at a relatively lower level.

Abstract: A method for identifying signals of the same origin is described. The signals are identified based upon a method of system analysis, wherein an accumulated value of an attenuation weighted by power of an incoming channel, the so-called correspondence coefficient, is used as the criterion whether two signals are of the same origin.

Abstract: Example methods, apparatus and articles of manufacture to cancel echo for a communication path having long bulk delays are disclosed. A disclosed example method includes determining a first location of a first coefficient having a largest magnitude of a first plurality of magnitudes associated with a first plurality of coefficients of a first phase; determining a second location of a second coefficient having a largest magnitude of a second plurality of magnitudes associated with a second plurality of coefficients of a second phase different than the first phase; comparing a difference between the first and second locations to a threshold; and, when the difference is less than the threshold, selecting a first offset based on a greater of the magnitude of the first coefficient and the magnitude of the second coefficient; and cancelling an echo contained in a signal using the first offset.

Abstract: A system includes a loudspeaker, a microphone, an echo canceling circuit, and an accelerometer coupled to the loudspeaker for providing a reference signal, a(n), to the echo canceling circuit. Preferably, the accelerometer is attached to the loudspeaker. A speaker signal, x(n), is used to drive the loudspeaker and the system also includes a switch for coupling either a(n) or x(n) to the echo canceling circuit, depending upon the volume or loudness of the sounds involved.

Abstract: An exemplary embodiment of the present invention is directed toward a method and system for cancelling line echo in the presence of a known secondary audio signal. Filter adaptation is enabled in the presence of a known secondary audio source such as the sound of a computer game, a music signal or other secondary audio sources that would otherwise prevent echo cancellation due to an apparent double talk condition. It is emphasized that this abstract is provided to comply with the rules requiring an abstract which will allow a searcher or other reader to quickly ascertain the subject matter of the technical disclosure. It is submitted with the understanding that it will not be used to interpret or limit the scope or the meaning of the claims.

Abstract: An input/output apparatus includes a speaker, a microphone, an adaptive filter circuit and a control circuit. The speaker is configured to output audio based on a voice-receiving signal. The microphone is configured to produce a voice-transmitting signal based on gathered audio. The adaptive filter circuit is configured to produce a residual signal based on a pseudo echo signal and a amplified voice-transmitting signal. The pseudo echo signal is produced from the voice-receiving signal and a transfer coefficient which expresses audio transfer properties between the speaker and the microphone. The amplified voice-transmitting signal is produced by causing a microphone amplification circuit to amplify the voice-transmitting signal. The control circuit is configured to stop output of the residual signal from the adaptive filter circuit during at least one of the speaker and the microphone is muted.

Abstract: A signal processing system which discriminates between voice signals and data signals modulated by a voiceband carrier. The signal processing system includes a voice exchange, a data exchange and a call discriminator. The voice exchange is capable of exchanging voice signals between a switched circuit network and a packet based network. The signal processing system also includes a data exchange capable of exchanging data signals modulated by a voiceband carrier on the switched circuit network with unmodulated data signal packets on the packet based network. The data exchange is performed by demodulating data signals from the switched circuit network for transmission on the packet based network, and modulating data signal packets from the packet based network for transmission on the switched circuit network. The call discriminator is used to selectively enable the voice exchange and data exchange.

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: A communications device that is configured to detect double talk is described. An echo canceller is configured to cancel an echo from an input signal using an adaptive filter. A double-talk detector provides a double-talk statistic. The double-talk statistic is proportional to the ratio of the remaining echo energy in the cancellation error signal and the total cancellation error energy.

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: 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 telecommunication system including a speakerphone provides a coupled signal path, including a microphone, configured to sense an incoming audio signal and an echo signal and generate a coupled signal. The echo signal includes non-linear distortion generated by a speaker of the speakerphone. An echo signal path, including an amplifier, is configured to sense the echo signal and generate an echo reference signal. The echo reference signal includes the non-linear distortion. An acoustic echo canceller is configured to receive the coupled signal from the coupled signal path, to receive the echo reference signal from the echo signal path, and to cancel out the non-linear distortion included in the coupled signal based on the non-linear distortion included in the echo reference signal.

Abstract: An echo cancellation device relies on the known characteristics of the sync frame to monitor, update in an off-line fashion and determine the accuracy of an echo canceller in, for example, a modem, such as an ADSL modem. Specifically, time domain samples are read from the transmit (Tx) and receive (Rx) paths of the modem. These samples are stored in memory. When the sync frame has received a predetermined number of the same Tx samples and Rx samples, the samples are stored. Running averages, over the sync frames, of the TX and RX samples are maintained. These averages are subtracted from a sync frame of samples, to allow LMS updating of the echo canceller taps, free of extraneous signals. Updating, i.e., tracking of changes in the echo channel, is done for the echo canceller in an off-line fashion. The coefficients for the in-line version are updated, while the off-line version is updated over several sync frames. Periodically, the performance of the off-line version is compared with the in-line version.

Abstract: Received data is filtered to produce pre-noise suppression data. Noise is removed from the pre-noise suppression data to provide noise-suppressed data. At least one weighted filter coefficient is dynamically determined using at least the pre-noise suppression data and not the noise suppressed data. The determination occurs independently from and is not affected by removing the noise from the pre-noise suppression data. Removing the noise from the pre-noise suppression data occurs independently from and is not affected by dynamically determining the at least one weighted coefficient.

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: Method and arrangement in a Vector Control Entity for efficient selection of which crosstalk to cancel when applying partial vectoring in a DSL communication system. The method comprises calculating a respective crosstalk effect indicator, CEIij, for the crosstalk from each line j of a plurality J of lines in a vectoring group, to a plurality of tones Si of a line i in the vectoring group, based on the strength of the crosstalk from each line j to line i, and thus providing one CEIi-value, associated with the plurality Sj of tones, per line j. The method further comprises allocating partial-vectoring resources to cancellation of crosstalk, based on the calculated CEIs. The arrangement is adapted to enable the performance of the above described method.

Abstract: Example methods, apparatus and articles of manufacture to cancel echo for communication paths having long bulk delays are disclosed. A disclosed example method includes determining a first location of a first largest magnitude of a first plurality of coefficients of an echo canceller, the first plurality of coefficients separated by two or more sample intervals, and cancelling an echo contained in a received signal using a second plurality of coefficients of the echo canceller and a first offset selected based on the first location, the second plurality of coefficients separated by one sample interval.

Abstract: The proposed echo canceller comprises:—an adaptive filter for receiving a signal affected by an echo, and for supplying a filtered signal that is an estimate of the echo;—a subtractor for subtracting this estimate from the received signal and supplying a residual signal;—means for detecting (7, 8) a ring back tone in said residual signal;—means for blocking (9) the received signal and replacing it by a locally generated ring back tone if a ring back tone is detected in the received signal;—a timer to determine a time period;—and means (11-17) for, during said time period, replacing the residual signal by some synthetic comfort noise when there is no ring back tone.

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

Abstract: An echo canceller includes an initial delay estimator producing a reference signal containing an identified frequency component masked by means of frequency masking effect to output the reference signal on an echo path, and detecting, from a sending voice signal, an echo component of the identified frequency component derived from the reference signal. The estimator finds an initial delay amount on the echo path on the basis of a forming timing of the reference signal containing the identified frequency component and a detection timing of the echo component in the identified frequency component in the sending voice signal. The echo canceller includes a delay circuit for delaying a received voice signal by the initial delay amount on the echo path obtained from the estimator to supply a resulting signal to a pseudo echo generator.

Abstract: An exemplary embodiment of the present invention is directed toward a method and system for cancelling line echo in the presence of a known secondary audio signal. Filter adaption is enabled in the presence of a known secondary audio source such as the sound of a computer game, a music signal or other secondary audio sources that would otherwise prevent echo cancellation due to an apparent double talk condition. It is emphasized that this abstract is provided to comply with the rules requiring an abstract which will allow a searcher or other reader to quickly ascertain the subject matter of the technical disclosure. It is submitted with the understanding that it will not be used to interpret or limit the scope or the meaning of the claims.

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 circuit includes a microphone circuit, an audio processing module, a digital audio processing module, and an active noise reduction (ANR) circuit. The microphone circuit receives acoustic vibrations and generates an audio signal therefrom. The audio processing module generates a representation of the audio signal. The digital audio processing module compensates the representation of the audio signal based on hearing compensation data to produce a hearing compensated audio signal. The ANR circuit receives the hearing compensated audio signal and an ANR signal. The ANR circuit further functions to adjust the hearing compensated audio signal based on the ANR signal to produce an output audio signal, wherein the ANR signal is generated based on the output audio signal.

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: In one embodiment, the present invention is a method for detecting an echo path change (EPC) in a telecommunications network. The method detects whether the effectiveness of echo cancellation of an echo canceller has decreased relatively significantly. Once a relatively significant decrease is detected, the method determines whether the decrease was an EPC or an inadvertent detection of double talk (DT). In particular, the method considers whether echo is effectively cancelled over a hangover period. Further, echo return loss (ERL) estimates are generated over the hangover period and compared to a lowest-possible ERL for the network. If both (1) echo cancellation is ineffective and (2) a sufficient number of ERL estimates are not below the worst-case ERL, then an EPC decision is made. If either (1) echo cancellation is effective or (2) a sufficient number of ERL estimates are below the worst-case ERL, then a DT decision is made.

Abstract: An input/output apparatus includes a speaker, a microphone, an adaptive filter circuit and a control circuit. The speaker is configured to output audio based on a voice-receiving signal. The microphone is configured to produce a voice-transmitting signal based on gathered audio. The adaptive filter circuit is configured to produce a residual signal based on a pseudo echo signal and a amplified voice-transmitting signal. The pseudo echo signal is produced from the voice-receiving signal and a transfer coefficient which expresses audio transfer properties between the speaker and the microphone. The amplified voice-transmitting signal is produced by causing a microphone amplification circuit to amplify the voice-transmitting signal. The control circuit is configured to stop output of the residual signal from the adaptive filter circuit during at least one of the speaker and the microphone is muted.