Abstract: An echo reduction system includes a signal analysis filter that converts an input into sub-band signals. A signal down-sampling circuit down-samples the sub-band signals at a first down-sampling rate. An echo analysis filter converts a loudspeaker signal into echo sub-band signal that are further processed by an echo down-sampling circuit. The circuit down-samples the echo sub-band signals at a second down-sampling rate to generate down-sampled echo sub-band signals. An echo compensation filter folds the down-sampled echo sub-band signals with an estimated impulse response of a loudspeaker-room-input system. A second echo down-sampling circuit down-samples the folded down-sampled echo sub-band signals at a third down-sampling rate to generate estimated echo sub-band signals. The first down-sampling rate is equal to the product of the second and third down-sampling rates.

Abstract: A voice processing apparatus includes a band dividing portion dividing a first voice signal generated by a first microphone and a second voice signal generated by a second microphone into predetermined frequency bands, a sound source segregating portion segregating an echo component of a voice emitted by a first sound source included in a voice emitted by a second sound source in each of the predetermined frequency bands based on the power of the first and second microphones, and a band synthesis portion synthesizing the first and second voice signals from which the echo component of the first sound source has been segregated by the sound source segregating portion into a voice signal including the voice emitted by the first sound source and a voice signal including the echo component of the first sound source.

Abstract: Systems and methods for blind subband acoustic echo cancellation postfiltering in a communication device are provided. In exemplary embodiments, an acoustic signal is received via a microphone of the communication device. Acoustic echo cancellation (AEC) is applied to the acoustic signal to obtain an AEC masked signal. Because residual echo may still exist in the AEC masked signal, blind subband AEC postfiltering on the AEC masked signal may be performed to obtain an echo-free acoustic signal. The echo-free signal may then be output.

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 multichannel echo canceller according to the present invention includes an echo cancellation section which receives loudspeaker input signals sp1, sp2 including a second acoustic signal to be respectively inputted to a plurality of loudspeakers 10, 20 provided in a first location and detection signals m1, m2 detected by a plurality of microphones 11, 21 provided in the first location; separates a first acoustic signal and a second acoustic signal, which are included in the detection signals m1, m2, by performing signal processing based on an independent component analysis; cancels as an echo the second acoustic signal, which is included in the detection signals m1, m2, by outputting only the separated first acoustic signal to a plurality of loudspeakers 30, 40 provided in a second location.

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: Methods and corresponding systems for reducing an echo signal include assigning a subsegment delay to each of a plurality of subsegment adaptive filters. A send signal and a delayed receive signal are coupled to each of the subsegment adaptive filters, wherein the delayed receive signal is delayed according to a respective subsegment delay. A set of filter coefficients in each of the subsegment adaptive filters are adapted, in parallel, to correspond to a respective subsegment impulse response of a connected system. Each set of filter coefficients is analyzed to produce a pure delay, and the pure delay is used to delay the receive signal for a main adaptive filter. An echo signal replica is produced using the main adaptive filter and an error signal. The echo signal replica is subtracted from the send signal to produce the error 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: A multi-channel audio decoder provides a reduced complexity processing to reconstruct multi-channel audio from an encoded bitstream in which the multi-channel audio is represented as a coded subset of the channels along with a complex channel correlation matrix parameterization. The decoder translates the complex channel correlation matrix parameterization to a real transform that satisfies the magnitude of the complex channel correlation matrix. The multi-channel audio is derived from the coded subset of channels via channel extension processing using a real value effect signal and real number scaling.

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: 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: An acoustic echo canceller (AEC) apparatus for canceling an echo signal input to a microphone in a mobile terminal. In the AEC apparatus, a first analysis filter bank divides a signal received from the microphone into a plurality of subband signals, and classifies the subband signals into high-band signals and low-band signals according to a predetermined reference. A second analysis filter bank divides a far-end user's signal into a plurality of subband signals, and outputs particular subband signals being lower than or equal to a reference band to an adaptive filter bank. The adaptive filter bank receives the subband signals output from the second analysis filter bank, and generates estimated echo signals for the individual subbands. An adder calculates differences between the low-band signals among the subband signals output from the first analysis filter bank and the estimated echo signals output from the adaptive filter bank.

Abstract: A system and process for dereverberation of multi-channel audio streams is presented which uses reverberation suppression techniques. In general, the present system and process builds a frequency dependent model of the reverberation decay and uses spectral subtraction-based reverberation reduction to achieve the aforementioned suppression. This dereverberation system and process can be used to improve automatic speech recognition (ASR) results with minimal CPU overhead.

Abstract: The present invention relates to an audio communication system and method with improved acoustic characteristics. It combines the benefits from full band echo cancellers and sub-band echo cancellers, by merging the filter taps of sub-band filters into the respective full-band filter taps, which is being used to generate the inverted echo estimate in the echo canceller.

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: A noise reduction device including: an input signal spectrum obtaining unit that obtains an input signal spectrum by a subband unit based on a current frame of an input signal; an averaged spectrum obtaining unit that obtains an averaged spectrum of the input signal by averaging the input signal spectrum; an estimated noise spectrum obtaining unit that obtains an estimated noise spectrum estimated based on a past frame of the input signal by the subband unit; and an SN ratio obtaining unit that obtains an SN ratio by the subband unit, based on the averaged spectrum of the input signal obtained by the averaged spectrum obtaining unit, the estimated noise spectrum obtained by the estimated noise spectrum obtaining unit, and a function of the averaged spectrum of the input signal obtained by the averaged spectrum obtaining unit and the estimated noise spectrum obtained by the estimated noise spectrum obtaining unit.

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

Abstract: A system such as a speakerphone may include a processor, memory, a speaker and a microphone. The processor may be configured (via program instructions stored in the memory) to calibrate the speaker by: outputting a stimulus signal; receiving an input signal corresponding to the stimulus signal; computing a midrange sensitivity and a lowpass sensitivity for a transfer function derived from a spectrum of the input signal and a spectrum of the output signal; subtracting the midrange sensitivity from the lowpass sensitivity to obtain a speaker-related sensitivity; performing an iterative search for current parameters of a speaker model using the input signal spectrum, the stimulus signal spectrum and the speaker-related sensitivity; and updating averages of the speaker model parameters using the current parameter values. The stimulus signal may be transmitted during periods of silence in the external environment. The parameter averages may be used to perform echo cancellation.

Abstract: A method of determining residue-echo threshold for the trans-hybrid echo path of a loop start (LS) trunk line, comprising the steps of estimating the impulse response of the trans-hybrid echo path, passing a speech-like signal through the LS trunk circuit and collecting the echo signal. The collected echo signal is compared to the convolution of the speech-like signal with the estimated impulse response. The difference between the convolution result and the collected echo signal is the residue echo, which represents the effect of nonlinearity. With different levels of excitation signal, a curve of residue echo power level against input signal level is obtained, and the distortion characteristics are derived from this curve.

Abstract: In a hands-free mode of speech communication, a loudspeaker produces acoustic energy of a distant signal from a far-end talker and a microphone produces a near-end signal containing a speech component representing the speech activity of a near-end talker or an acoustic echo component, or both. An echo replica is produced from the distant signal and a residual echo representing the difference between the near-end signal and the echo replica. The residual echo is used as a feedback signal to produce the echo replica. Using one of the near-end signal and the residual echo as a first input signal and the echo replica as a second signal, an estimate of the acoustic echo is produced when the speech activity of the near-end talker is low or zero. Using the acoustic echo estimate, the spectrum of the first input signal is shaped to produce a local signal for transmission to the far-end talker.

Abstract: Disclosed herein is a three-way call detection system and method for detecting the addition of a third party to a pre-existing telephonic connection between a first party and a second party. The system includes a meter for measuring the amplitude of a line voltage of a telephonic connection between two parties, a first filter to remove a frequency band from the line voltage, a white noise generator for generating a low level white noise signal in the removed frequency band, a coupler for combining the white noise signal with the filtered signal, to create an additive line voltage, a second filter that passes only a portion of the additive line voltage including at least a portion of the white noise signal, and a processor for monitoring the passed signal for a shift in amplitude level associated with the addition of a third party to the connection.

Abstract: A system for canceling acoustic feedback includes an input for receiving a digital audio signal and a processor configured to detect acoustic feedback signal in the digital audio signal and to determine the frequency of the feedback signal The system also includes a plurality of bandpass filters for attenuating the feedback signal. The processor is further configured to: select a bandpass filter from among the plurality of bandpass filters. The selected bandpass filter comprises a response characteristic that attenuates parts of the signal at the frequency of acoustic feedback signal.

Abstract: A novel technique for canceling echo signal in teleconferencing applications. In one example embodiment, a low complexity double-talk and noise robust frequency domain adaptive filter is used to cancel the echo signal during the teleconferencing applications. The adaptive filter computes step sizes using power spectral density functions of error and far end signals using an equation, which does not require a single talk/double detectors and a voice activity detector. The computed step sizes are then used by the adaptive filter (PBFDAF) to cancel the echo signal.

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

Abstract: A method and apparatus for adapting dual filters is disclosed. In one aspect, a method may include transforming a signal, adapting a first adaptive filter based on the transformed signal, estimating a delay of an impulse response based on the adaptation of the first filter, delaying a signal based on the estimated delay, and adapting a second adaptive filter based on the delayed signal. In one aspect, an echo or other unwanted signal may be reduced or cancelled based on the adaptation of the second filter.

Abstract: 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 apparatus and method for canceling an echo signal is disclosed. The apparatus includes an echo filter, an up-sampler, an interpolation filter, and a signal combining apparatus. The echo filter filters the transmitted signal, the up-sampler up-samples the output of the echo filter, the interpolation filter filters the up-sampled output to generate a signal that emulates the echo signal, and the signal combining apparatus combines the signal emulating the echo signal with the received signal to remove echo signal. The method estimates an impulse response of the echo channel for estimation of filter coefficients of an echo filter. The transmit signal is then filtered in the echo filter to generate an output, which is up-sampled to generate another output. The up-sampled output is filtered by the interpolation filter to generate a signal that emulates the echo signal, which is then combined with the received signal to cancel the echo signal.

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

Abstract: A method, apparatus, system, and signal-bearing medium that in an embodiment determine a degree of correlation between a speaker output signal and a microphone input signal and modulate an adaptive gain of an acoustic echo canceller based on the degree of correlation.

Abstract: Methods and systems for a mixed time-frequency domain compander, where the expander and compressor are applied in the frequency domain (subbands), while the limiter and the linear gain are applied in the time-domain (full band). Far-end noise reduction is achieved by adapting the expander gain to the far-end noise, near-end noise compensation is achieved by adapting the linear gain and the compressor gain to the near-end noise. The range of the linear gain section is set individually for each subband, but the linear gain is essentially applied to the full band signal via a scaling procedure. The scaling procedure first reduces the compander gain in the frequency domain by a scaling factor and then performs the reciprocal reverse scaling in the time domain. Far-end noise reduction is coupled to near-end noise compensation in order to avoid cross-modulation of the far-end noise by the near-end noise.

Abstract: A howling control apparatus and howling control method for controlling time up to the cancellation of howling suppression according to a howling occurrence situation, thereby eliminating the repetition of howling suppression and cancellation. A howling detecting section (104) detects howling based on the band level and the band level average value and measures time for which no howling occurs, a waiting time setting section (105) decides waiting time to be set this time from time for which no howling occurs and a previous waiting time, and a gain control section (106) causes a gain that is set to a howling suppressing section (107) to be retuned to a normal value during the waiting time.

Abstract: Disclosed herein is a three-way call detection system and method for detecting the addition of a third party to a pre-existing telephonic connection between a first party and a second party.

Abstract: A transform-domain adaptive filter uses selective partial updating of adaptive filter parameters. This updating may be based on a constrained minimization problem. The adaptive filter parameters are separated into subsets, and a subset is selected to be updated at each iteration. A normalization process applied to the frequency bins prior to multiplication by the adaptive filter parameters is used to prevent adaptive filter lock-up that may be experienced in the event of high energy levels of signals in particular frequency bins. Convergence of the transform domain filter is ensured at a rate generally faster than a corresponding time-domain adaptive filter. The transform-domain adaptive filter may be used for various applications, including system identification, channel equalization, or echo cancellation.

Abstract: A method and an apparatus for selecting an antenna from an antenna diversity array in a wireless communication system are provided. The method includes the steps of modulating an incoming signal with a spread-spectrum type modulation, measuring a signal quality value for each antenna in the antenna diversity array, and selecting an antenna from the antenna diversity array on the basis of the measured signal quality values. The step of measuring a signal quality value for each antenna may include computing peak-to-average ratio values by dividing a peak sample value by an average sample value, where the average sample value is determined by averaging all of the predetermined number of sample values for each of the plurality of incoming spreading codewords, and the peak sample value is determined by selecting the maximum of all of the predetermined number of sample values for each of the plurality of incoming spreading codewords.

Abstract: Systems and methods for reducing transmit echo in a transceiver are disclosed. A hybrid echo canceller includes a limited tap length FIR filter to cancel short-term echo, while an interpolated filter is used to cancel the long-tail echo. The limited tap length FIR filter adapts and calculates coefficients for each adapted tap. Taps of the interpolated filter, on the other hand, are adapted but coefficients are calculated for a subset of the taps. Various interpolation schemes may be applied to the calculated coefficients to associate a coefficient with each tap of the interpolated filter. The technique presented produces an effective filter length of N taps with a reduction in computation and signal processing resources. Preferred embodiments of the echo canceller may be construed as methods for reducing transmit echo.

Abstract: A multichannel full-duplex audio signal transmission system, comprising an adaptive filter (2) provided for multichannel acoustic echo cancellation. A channel combining device (5) is provided between the preprocessing units (V1, . . . VD) and loudspeakers (L1, . . . ,LD), in which several (C>D) loudspeakers can be connected to one and the same preprocessing unit and by means of which the remaining D-C preprocessing units can be separated from the loudspeakers (L1, . . , LD). The channel combining device permits an optimization of the convergence ratio of the adaptive adjustment of filter coefficients in said adaptive filter (2).

Abstract: A method of controlling an echo canceller in a communications channel, is disclosed wherein input signals from the communications channel are first subbanded into a subband. Echo locations are then identified within the subband and are used to control the echo canceller. Typically, the echo canceller will be a fullband echo canceller with an adaptive filter, in which case the echo locations are used to control the filter coefficients.

Abstract: Disclosed is an apparatus for cancelling far endpoint echo signals in audio signals transmitted from a near endpoint to a far endpoint. In one embodiment, the apparatus includes a near endpoint analysis filter bank operable to divide a near endpoint signal into a plurality of near endpoint subband signals, a far endpoint analysis filter bank operable to divide a far endpoint signal into a plurality of far endpoint subband signals, and a background signal power estimator operable to determine background noise at the near end.

Abstract: Disclosed herein is a three-way call detection system and method for detecting the addition of a third party to a pre-existing telephonic connection between a first party and a second party. The system comprises a meter for measuring the amplitude of a line voltage of a telephonic connection between a first party and a second party.

Abstract: An echo canceler for canceling echo components having a plurality of cascade-connected adaptive filters and a plurality of update control means. The echo canceler may further include an update control selective circuit for generating selective signals which enable or disable the update control means. Power values of opposite speech signals generated from the adaptive filters are used to control the updating of filter coefficients of the adaptive filters.

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

Abstract: The present invention relates to a method for improving acoustic noise attenuation, in which the method includes an adaptive control for partial-band echo compensation as well as global-band post-filtering for suppressing residual echo in hand-free devices. This method also uses a level balance as well as controllable frequency-selective echo compensation with partial-band processing. After the frequency-selective echo compensation, the signal is submitted to a post-filtering in another frequency-selective filter using a Wiener-equation adjustment algorithm (Wiener filtering). A single control value (increment vector) is used for controlling both the frequency-selective echo compensation and the other filter. This method can thus be implemented with a very reduced amount of calculations so that it can also be used in simple consumer-directed processors.

Abstract: A base band echo cancellation method and apparatus utilizing Laguerre echo estimation. The method begins with a series of transmitted symbols that are stored in a buffer. The buffer is segmented into a head portion and a tail portion. The symbols in the tail portion are passed through a Laguerre or IIR filter and then reinserted into the tail portion of the buffer. An echo estimation signal is generated based on the symbols in the buffer. The apparatus includes a transmit symbol buffer that is coupled through a modulator to a Laguerre or IIR filter. A demodulator couples the filter back to the buffer. An echo estimator is also coupled to the buffer.

Abstract: Echo-canceling apparatus of the invention includes transfer function estimation unit which estimates a transfer function corresponding to the reverberation of a room attached to a voice after it is output from a loudspeaker and before it is input to a microphone, a first filter unit which operates using the transfer function, a first subtraction unit which subtracts the output signal of the first filter unit from the signal from the microphone, a second filter unit which operates using the transfer function copied from the first filter unit in case the estimation accuracy of the transfer function estimation unit is high, a second subtraction unit which subtracts the output signal of the second filter unit from the signal from the microphone, a singing detection unit which detects singing, a notch filter unit which notches a specific frequency band component in the signal received from a far-end speaker, and a switch unit which selects between the signal from the far-end speaker processed by the notch filter u

Abstract: An echo canceller processing echo, noise and near end talk in a narrower, but still intelligible, frequency band for reducing required processing power and complexity. In a preferred embodiment of the present invention, an input audio signal of captured sound in an audio communication system is decimated and then divided into a number of sub bands by an analyze filter. Each sub band is processed as in background audio echo cancelling by subtracting the signal with an echo estimate from a model of the acoustic signal in the respective sub band, except from that the signal is also bypassed, adjusted by a filter and subtracted from the processed signal. The resulting signals are then recombined by a synthesize filter and interpolated to the original sampling rate and bandwidth. Finally, the output from the synthesize filter is added to the input audio signal, which has been delayed and adjusted by a filter.

Abstract: A method and system for improving performance of an echo canceller with low additional complexity. Specifically, the present invention discloses using information from internal variables of Code Excited Linear Prediction (CELP) based codecs in a digital communication network to significantly improve the rate of convergence of the echo canceller. In one embodiment of the present invention, an error signal associated with a voice signal is filtered by a transversal filter using Linear Predictive Coefficients (LPC) coefficients to provide filter transfer functions for a Filtered-X Least Mean Squares algorithm. Additionally, an adaptive filter applies the Filtered-X Least Mean Squares algorithm using the pre-filtered voice signal available in a CELP-based decoder to create a synthetic echo signal which is subtracted from the echo signal for attenuation.

Abstract: A system for processing audio signals in two channels, wherein each channel includes a plurality of band pass filters for dividing the signal in each channel into a plurality of bands and a plurality of variable gain amplifiers coupled one each to a band pass filter. Control logic coupled to the variable gain amplifiers prevents the signals in adjacent bands within a channel from being attenuated simultaneously. The control logic attenuates the signals in one channel in accordance with the signal in the corresponding bands in the other channel. In some circumstances, the signal in the band in the one channel is attenuated.

Abstract: A double-talk detector (DTD) method of performing double-talk detection, an echo canceller, and a method of performing echo cancellation is used with an echo canceller (EC) to sense when an echo is corrupted by near-end speech (NES). The double-talk detector inhibits the adaptation of a synthesizing filter when NES is present, in order to avoid divergence of the adaptive algorithm. Due to the time varying properties of the echo path and the signal levels, a suitable decision threshold ensures the accuracy of the DTD. The double-talk detector utilizes an adaptive decision threshold which is capable of tracking variations in the echo path and signal/noise levels during a call.