Abstract: Apparatus and accompanying methods for use in a long-haul echo control device to automatically measure sufficient changes in path length and compensate a long-haul echo canceller for resulting variations in echo occurring over a path. The apparatus measures a round-trip path delay interval of, e.g., a predefined signaling tone(s) sent between international switching centers while establishing an outbound call connection over the path. Once the delay is measured, the apparatus sets echo delay of a corresponding long-haul echo canceller to accommodate that delay. For a multi-channel facility (e.g., T1 or E1), path delay is separately measured for each channel on that facility, and then the apparatus changes the echo delay to the same amount for the corresponding echo canceller associated with each and every channel on that facility.

Abstract: Long distance telephone connections must commonly employ echo cancellers to eliminate irritating echo. The normal practice is to additionally employ non-linear processors (NLPs) as an integral component of an echo canceller in order to eliminate small residual echo which remains after initial echo cancellation. While such devices do indeed remove remaining residual echo, they also may introduce additional problems, such as suppression of the background signal. However, in some cases there is no need to deploy a potentially problematic non-linear processor. If the overall route delay of the telephone connection is not excessive, the participants do not notice the residual echo. Therefore, a determination is made of the end to end delay of the overall telephone connection. If the determined delay is not excessive, non-linear processing is not applied or is applied less aggressively. This results in a clearer telephone call, free from non-linear processor problems.

Abstract: An echo cancellation invention which does away with the need for an accurate double-talk detector, while maintaining a higher adaptation gain for quicker convergence and also providing increased stability. It does this by operating two filter models of the acoustic path in parallel. One model is adapted continuously to find the most accurate model of the echo path, while the other is not. The echo canceller output is taken from the filter that is not adapted. A comparison of the residual error (echo) is done between the model being adapted and the model being listened to at regular intervals. When the model being adapted has less error (echo) than the model being listened to its filter coefficients are copied to the other model. If the model being adapted has greater error (echo) than the other model (caused by noise diverging the adaptation process) then the adapted model has its coefficients overwritten by the filter that is listened to.

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: A method and apparatus enables echo reduction in a full duplex transceiver system. A replica current is subtracted from a receiver via a first differential circuit path that adaptively matches a time constant associated with a second differential circuit path that connects the receiver with an external data line.

Abstract: A method and apparatus is capable of detecting the presence of a non-linear echo path as well as transitions from non-linear echo path to linear echo path and vice versa. Additionally, when non-linear echo path is detected, the echo canceller (12) is configured to follow a prescribed logic routine that combats possible echo leak-through as a result of compromised cancellation due to non-linearity. This prescribed logic routine affords the echo canceller (12) protection against non-linear echo path by preserving an already converged state of the echo canceller (12) from possible divergence due to the occurrence of non-linearity in the echo path and thereby prevents echo leak-through that would occur due to divergence of the echo canceller (12).

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: An echo canceler has its otherwise conventional tapped delay line basis functions augmented with at least one pair of basis functions realizing two quadrature responses of a complex pole. The frequency of this complex pole is selected as substantially equal to the frequency of the dominant pole of a highpass filter inserted into the echo path of the echo canceler, where the dominant pole is the pole closest to the unit circle. This echo canceler arrangement results in a substantial improvement in attainable echo-return-loss enhancement (ERLE).

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: An echo canceller and a method of echo cancellation using an NLMS algorithm. The NLMS algorithm alternates adaptation between a slow mode, aggressive mode, and inhibition mode, according to the status of the echo canceller. This implementation requires no correlation information. This implementation is simple and not very time consuming. During initial conversation, the NLMS algorithm is in an aggressive mode to ensure fast convergence. After convergence, the NLMS algorithm switches to a slow mode for lower residual echo return. Whenever an abrupt increase of residual echo caused by either double talk or echo path change is detected, the current adaptive filter coefficients are retained and then updated in the aggressive mode. The decision as to which echo estimation (the response of the retained filter or the response of the aggressively adapted filter) is subtracted from the reference echo for echo cancellation within a short detection delay period is determined according to the Geigel algorithm.

Abstract: A two-step training method for the estimation filter in the echo cancellation (EC) path of the analog front-end (AFE) circuit for a modem, such as an asymmetric digital subscriber line (ADSL) modem. During the first step, the transmit (TX) path is turned off and a digital TX signal (e.g., a pseudo white gaussian noise signal) is applied to the EC path essentially bypassing the EC estimation filter. During the second step, the EC path is turned off and a digital TX signal (e.g., the same pseudo white gaussian noise signal) is applied to the TX path. Coefficients for the EC estimation filter are then generated based on the digital RX signals recorded during the two training steps. In particular, the recorded digital RX signals are transformed to the frequency domain, where one is divided into the other. The resulting digital signal is then transformed back to the time domain, where it is used to generate the coefficients for the EC estimation filter.

Abstract: An echo canceling device includes a first variable amplifier for variably amplifying an audio signal input through a microphone according to a control signal; a first analog-to-digital converter for converting an analog signal output from the first variable amplifier to a digital signal; a second digital-to-analog converter for converting a digital signal output from a digital signal processor to an analog signal; a second variable amplifier for variably amplifying a signal output from the second digital-to-analog converter according to a control signal; an input/output connector for connecting a signal output from the second variable amplifier to internal and external processing devices; a sound measurer for measuring a level of surrounding sounds input through the microphone; and the digital signal processor for executing an echo canceling routine, performing general digital signal processing and generating the control signals for controlling the first and second variable amplifiers according to the sound l

Abstract: A method and apparatus for performing double-talk detection in an acoustic echo canceller in which a detection statistic is advantageously computed based on an estimate of a cross-correlation between the far-end signal and the return signal which has been normalized with use of an estimate of a covariance matrix of the far-end signal. The estimate of the cross-correlation between the far-end signal and the return signal may be further normalized with use of either an estimate of a variance of the return signal or an estimate of a covariance matrix of the return signal. In certain illustrative embodiments of the invention, one or more of these quantities may be estimated based on signal samples sampled over a predetermined time window. And in another illustrative embodiment of the present invention, the coefficients of the adaptive filter employed in the acoustic echo canceller itself are advantageously used to compute the detection statistic.

Abstract: A method of echo cancellation for a signal transmission system in which the size of the step used to adapt filter coefficients is adjusted in accordance with the echo delay and in which stationary signals are avoided by determining when the mean and variance of coefficients obtained from a second order linear predictive coding analysis of successive far end samples exceed preset thresholds.

Abstract: An echo canceling apparatus includes a long-response echo canceler that removes an echo of an incoming signal from an outgoing signal by operating on a most recent segment of the incoming signal, stored in a tapped delay unit. The apparatus has a control unit that calculates a cancellation characteristic for each tap of the tapped delay unit. The cancellation characteristics are used to select a subset of taps corresponding to a changing component of the echo, the subset being empty if there is no changing component. A short-response echo canceler, operating on the outputs of the selected taps, cancels the changing component of the echo left uncanceled by the long-response echo canceler. Alternatively, the step gain of the selected taps is increased. The short-response echo canceler is preferably disabled in the double-talk state.

Abstract: Processing complexity in an acoustic echo canceller can be reduced by partitioning the input near-end and far-end signals into respective pluralities of frequency subband signals. Also, due to the complex conjugate symmetry of conventional Discrete Fourier Transform matrices, not all of the subband signals of each plurality need be subjected to echo cancellation processing. The subband signals that would have otherwise been produced by echo cancellation processing are instead provided by producing complex conjugates of selected ones of the subband signals that have been subjected to echo cancellation processing.

Abstract: Echoes caused by discontinuities in a telecommunications transmission lines cause double talk which cause divergence problems with echo cancellers. Divergence problems in echo cancellers cause considerable problems in interactive caller response systems, they cause less problems if all parties to a call are human. One of the biggest problems is the effect of the echoes on automatic speech recognition systems. The echo canceller method and system presented cancels echoes to reduce their interference with ASR and yet provides stability in the presence of double talk.

Abstract: A fault associated with a communication medium is analyzed by correlating a first set of information indicative of echo associated with the communication medium with a second set of information indicative of echo. The fault is characterized based on the correlation between the first set of information and the second set of information. The first set of information and the second set of information can include echo canceller coefficients generated by an echo canceller coupled to the communication medium.

Abstract: A speaker-phone that is operable to perform system change detection, double-talk detection and if desired, optimal step-size control. The invention differentiates between system change and double-talk and is operable to these two situations differently. The invention does not rely solely on the energy measurements that are employed by conventional speaker-phones. Highly reliable system change detection is offered when compared to conventional speaker-phones. The direction and the amplitude of the system change is measured with very good detection probability in the invention. For double-talk detection, the invention does not rely on assumptions that the far-end echoes are of lower amplitude that the near-end signal in performing double-talk detection. The invention performs double-talk detection within a large diversity of echo-paths and background noise levels. A designer may tune the speaker-phone system to ensure proper double-talk detection near low levels of the near-end signal.

Abstract: A method and apparatus for reducing echo feedback in a wireless communication system (10) is accomplished when a receiving communication unit (24) senses a feedback signal, or echo, via an ancillary communication path. The receiving communication unit is a targeted recipient of an original audio signal generated by a transmitting communication unit (22), where the original audio signal (42) is conveyed to the receiving communication unit. Upon detecting the feedback signal and determining that it exceeds a feedback threshold, the receiving communication unit attenuates an audible output of the original signal to reduce echo to the transmitting communication unit. In addition, the receiving communication unit, and/or the transmitting communication unit include echo canceller to further minimize the echo within the digital communication system.

Abstract: The present invention relates to an echo reduction method implemented with a multi-sensor sound pickup device (8) forming an antenna (10), and a sound playback device (13). The method consists in subjecting the output signals ym(t) from the sensors (11m) to complex weights w(f), said weights w(f) being calculated by maximizing the directivity factor Fd(f) under constraints at low frequencies and in the near field. The invention also provides an echo reduction apparatus (6). The apparatus comprises: a multi-sensor sound pickup device (8) forming an antenna (10); means for calculating a first constraint; and means for calculating a second constraint.

Abstract: The step sizes for tap positions of an adaptive filter (124) in an echo canceller (125) for canceling an echo signal are individually selected based upon a profile of the echo environment. A training signal (112) is injected into the echo canceller and to the echo path during a training interval, or period. The adaptive filter profiles the echo path during the training interval. Respective step sizes &agr;[k] for the tap positions are generated as a function of the echo profile so generated.

Abstract: Switch 141 continuously switches between received signal 2 and a supplemental signal, which is obtained by processing the received signal 2 through filter 145 to use the output of switch 141 in place of received signal 2. Accordingly, adaptive filters 122 and 124 operate sometimes by using received signal 2 as the input signal and sometimes by using the supplemental signal as the input signal, so that it is possible to obtain adaptive filter coefficients by using twice the number of conditional equations as the case of using only received signal 2 as the input signal. Therefore, since the adaptive filter coefficients do not becomes indefinite, it is possible to converge the coefficients to the correct values. Further, since switching period between the original and the supplemental signals is controlled to be longer than the sampling period of the received signal, it is possible to suppress aliasing distortion of the received signal directly supplied to a speaker and to be maintain better sound quality.

Abstract: One aspect of the invention relates to a method for line matching of analog communication lines in telecommunications networks, whereby signal processor(s) and/or programmable digital filters are used. Coefficients of the line transmission function are calculated by an echo compensator to be compared with pre-specified filter coefficients in a filter device assigned to the line, in which case, an incorrect line matching is identified and a new set of filter coefficients is generated, which is fed into the filter device. In this case, the first values to be accepted and processed for the coefficients of the line transmission function calculated by the echo compensator and used for comparison, are those that are determined by an echo compensator for a first line gateway in a line path, which is associated with the reflection with the shortest delay time. In networks with high dispersion of the line parameters, the invention re-establishes the task distribution between line matching and echo compensator.

Abstract: When an echo attenuation amount ACOM(i) is smaller than a desired echo attenuation amount ACOM, and an echo cancellation amount ACANC(i) is equal to or greater than a echo/noise ratio E/N(i), a block length computer increases the block length BL(i) and outputs that increased block length BL(i) to a block length controller. The echo canceler comprises such a block length computer and such a block length controller, wherefore the echo attenuation amount can ultimately be made to be equal to or greater than the desired echo attenuation amount.

Abstract: An apparatus and method are provided for predictive provisioning of functional packages based on offered traffic and a predictive model of the offered traffic. According to one embodiment of the invention, a managed agent provides predictions regarding an anticipated need for functional package provisioning based on traffic offered to the managed agent and a predictive model of offered traffic. A provisioning agent receives the predictions and in response to the predictions instructs the managed agent to provision a new functional package. According to another embodiment of the invention, the provisioning agent instructs a second managed agent to provision the media resource needed by the original managed agent. The traffic is then routed from the original man aged agent to the second managed agent.

Abstract: The renewal of a group of first tap coefficients is controlled according to a reception signal, a transmission signal and a first error signal. In contrast, the renewal of a group of second tap coefficients is controlled according to only the reception signal. An echo canceled output signal is produced from the group of first tap coefficients in cases where an estimate precision of the group of first tap coefficients is higher than that of the group of second tap coefficients, and an echo canceled output signal is produced from the group of second tap coefficients in cases of the before-initial estimate, the echo path change and the infinite return loss. Accordingly, the echo canceled output signal, in which the echo is canceled, can be reliably produced regardless of the echo path change or the infinite return loss.

Abstract: A method for canceling echo. A threshold value signal is set to a preset value and the threshold value signal and a medium value signal are keep at a level of the preset value. Diagnosis is made as to whether a difference between an echo signal (I) and the medium value signal (I-1) is more than a positive threshold value or less than a negative threshold value. When the difference between an echo signal (I) and the medium value signal (I-1) is more than the positive threshold value, the level of the medium value signal (I) is increased by an upraise speed and the level of the threshold value signal (I) is increased by a first speed faster than the upraise speed. When the difference between the echo signal (I) and the medium value signal (I-1) is less than the negative threshold value, the level of the medium value signal (I) is decreased by a drop speed and the level of the threshold value signal (I) is decreased by a second speed faster than the drop speed.

Abstract: An method and a device for echo suppression in a handsfree apparatus, particularly for a telephone is disclosed. With the aid of a control unit, control signals are generated from the signal of the far-end subscriber, the output signal of the loudspeaker-room-microphone system and the output signal of an adaptive filter used as reference, the control signals having an amplitude and a time response dependent on the speech mode. Using the control signals, the amplifications VE, VS of controllable amplifiers disposed in the receiving and transmitting paths are adapted to the speech mode, thus reducing the residual echo. The stepless transition of the amplifications from a quiescent value to a value attenuated with the factor and having an exponential time characteristic in response to a change of speech mode, provides an impression of natural speech.

Abstract: Disclosed is an echo path delay estimating apparatus and method for estimating the echo path delay using decimators and an adaptive filter, so that a short computation time is used by the echo canceller.

Abstract: An echo cancellation device (ECD) comprises an echo canceller (EC) including a transfer function estimator (EST, H) and a subtractor (ADD) and a residual echo suppression device (G, ADD2). The residual echo suppression device (G) comprises a residual echo filter (G) having an adjustable filter function (g). This filter function (g) can be adapted to either remove from the subtractor output (TNE′) the spectral characteristics relating to the reception signal (RFE) and/or to emphasize in the subtractor output signal (TNE′) a background signal spectral content relating to the transmission signal (TNE). A noise generation means (NGM′) can be provided at the output of the adaptable filter (G) for injecting a noise process in to the filter output signal (TNE′) prior to a speech coding in a speech coder (COD). The noise process masks in the filter output signal a spectral content relating to the reception signal (RFE).

Abstract: A circuit able to attenuate echo and side tones in a switching system connecting the first and second subscribers, comprises a first linear converter for converting pulse code modulated data generated by the first subscriber into an analog signal in the read message mode of the time switch, a side tone canceling filter for eliminating the side tones from the analog signal produced by the first linear converter, a clipper for clipping the analog signal rid of the side tones, a first PCM converter for converting the output signal of the clipper into a refined PCM data written into the time switch in the write message mode of the time switch and transferred to the second subscriber, a second linear converter for converting the PCM data generated by the second subscriber into an analog signal, an echo canceling filter for eliminating the echo from the analog signal produced by the second linear converter, and a second PCM converter for converting the analog signal rid of the echo into a refined PCM data written i

Abstract: An echo canceller (110) for a full-duplex communication system (100) such as a speakerphone determines a gain factor to be used in sub-band adapter coefficient updates by using both time domain and sub-band signals to determine the state of the system (100). In one embodiment a gain factor controller (120) uses the sampled time domain signal to determine if the system (100) is in the silence state, and if not uses the sub-band signals to determine whether the system (100) is in the talk, double-talk, or listen state. The gain factor controller (120) selectively changes the gain factor and selectively performs coefficient updating based on the state. In another embodiment the gain factor controller (120) determines the state and selectively updates coefficients using state information developed on a band-by-band basis.

Abstract: The magnitude processing circuit 607 non-linearly processes the received signal 2 and then outputs a non-linear processed signal. The delay processing circuit 300 further delays the non-linear processed signal and then outputs a delayed processed signal. The delayed processed signal is time-multiplexed with the non-linear processed signal. The multiplexed signal is used instead of the received signal 2. The adaptive filters 122 and 124 operate alternately with the non-linear processed signal or with the delayed processed signal obtained by non-linearly processing and then delaying the received signal 2 as the input signal. Adaptive filter coefficients can be obtained by using twice the number of conditions compared with the number is twice that for the case where only the received signal 2 is used as the input signal.

Abstract: The Least Mean Square (LMS) and Normalized LMS (NLMS) algorithms commonly employed in adaptive filters of echo cancelers are further optimized. Finite impulse response (FIR) filters are used to estimate a transfer function of an echo channel in a communications link. The LMS and NLMS algorithms are used to adapt the filter coefficients of the estimated transfer functions. By including the echo channel energy gain in the LMS or NLMS update equation, adaptation speed is increased by making adaptation responsive to the channel energy gain. An algorithm for estimating the echo channel energy gain adapts the estimate based on measured system parameters, such as a measured instantaneous channel gain and a near-end voice level. By considering in the NLMS algorithm, the average energy of either the microphone signal or the error signal, as well as the standard reference signal, a higher nominal update gain can be used. With a higher nominal update gain, the NLMS algorithm will converge more quickly.

Abstract: In methods and apparatus for echo cancellation, a noise estimate is selectively added to a downlink communications signals to improve the convergence speed and stability of an echo-canceling adaptive filter. The added signal content enables the echo-canceling adaptive filter to more quickly track echo path changes during user communications and prevents divergence of the adaptive filter during periods in which the downlink signal does not contain information (e.g., speech) sufficient to develop a quality echo path estimate. To prevent performance degradation from a user perspective, the added noise is made to resemble existing system noise (e.g., either near-end or far-end background noise). In exemplary embodiments, filter convergence speed and stability are further improved by whitening the spectrum and increasing the power level of the added noise whenever a near-end user is active and masking the added noise.

Abstract: Methods and apparatus for processing at least one voice signal in which a centralized voice processing unit controls operation of a plurality of voice processing block. In a first embodiment, the centralized voice processing unit comprises a centralized voice activity detector that provides at least one voice activity indication to the plurality of voice processing blocks. In a second embodiment, the centralized voice processing unit comprises a centralized noise estimator that provides at least one noise estimate to the plurality of voice processing blocks. In a third embodiment, the centralized voice processing unit comprises a centralized signal characteristic estimator that provides at least one signal characteristic estimate to the plurality of voice processing blocks.

Abstract: Methods and apparatus for compensating for the background noise modulation caused by the operation of a non-linear processor on an audio signal include a controller that receives operating status information from the non-linear processor and generates a noise reduction control signal. The methods and apparatus further include a noise reduction circuit that receives the noise reduction control signal from the controller. The noise reduction circuit reduces the background noise in the audio signal when a noise reduction control signal is generated by the controller and injects synthesized background noise into the audio signal when a noise reduction control signal is not generated by the controller.

Abstract: Methods and apparatus for processing a transmitted voice signal include a centralized frame controller providing at least one boundary control signal to voice processing blocks and controlling the operation of the voice processing blocks on the transmitted voice signal based upon the boundary control signal.

Abstract: An enhanced near-end voice signal may be generated in a hands-free environment by receiving an audio signal, generating an estimated acoustic echo signal, and generating a processed signal by removing the estimated acoustic echo signal from the audio signal. A near-end enhanced spectrum is then determined, that has one or more ranges of contiguous frequencies over which the detector spectrum takes on its largest values, wherein the range of contiguous frequencies are those associated with a relatively high echo return loss in the processed signal. The processed signal is filtered in accordance with the near-end enhanced spectrum, thereby generating an enhanced near-end voice signal. The enhanced near-end voice signal may then be applied to any of a number of components that are intended to process near-end speech. For example, when applied to a voice activity detector, the amount of energy contained in the enhanced near-end voice signal is then measured.

Abstract: A compander circuit (20) for echo cancellation in telecommunications apparatus with an estimator module (21) for estimating an echo coupling and a delay of a line echo and a characteristic module (23) with an associated multiplier (24) for controlling the short-time output level of zout(k) as a function of the short-time input level of zin(k) is characterized in that at all inputs and outputs of the compander circuit (20), a plurality of N communications channels are applied and outputted, respectively, at an N-fold clock rate as compared with a single communications channel; that all memory units in the compander circuit (20) are designed as multiplexer-buffer-demultiplexer (MBD) units (30); and that a modulo-N counter (26) is provided which controls all MBD units (30) so that in each of the MBD units (30), always the same memory cell (351, 352, 35N) assigned to a particular communications channel is addressed.

Abstract: Full acoustic coupling and echo cancellation is provided in implementations where there is acoustic coupling between acoustic zones.

Abstract: A process for suppressing total echoes comprising a plurality of partial echoes in telecommunications (TK) equipment such as terminals, transmission systems, or switching equipment by means of adaptive FIR (=Finite Impulse Response) filters that simulate the total echo and subtract it from the echo-affected useful signal transmitted to the respective TK equipment, whereby n partial FIR filters are provided, which respectively simulate a partial echo, is characterized in that m parallel, mutually independent TK channels are provided with an echo suppression function and that for this purpose a variable sub-quantity of the partial FIR filters is assigned to a TK channel to be operated, respectively, as required depending on the number of partial echoes and the size of the occurring delay times between signal and echo.

Abstract: An equipment (1) for the suppression of whole echoes composed of several partial echoes in telecommunications (TC) equipment, such as end devices, transmission systems or exchange equipment, with the help of adaptive FIR (=Finite Impulse Response) filters, which reproduce the whole echo and extract it from the echo-laden useful signal and which has been transferred to the respective TC equipment, in which the homogeneous FIR filters or partial filters (10), each of which are able to reproduce a whole echo or partial echo, are implemented on a single ASIC (=Application Specific Integrated Circuit) (9) for the m parallel and independent TC channels n to be served; and in which the whole echoes or partial echoes to be reproduced can be calculated in real time in the ASIC (9) and in which a digital signal processor (4) is designed to control the ASIC (9) and set the filter settings, especially after the coefficient required to reproduce the echo is calculated on the ASIC (9).

Abstract: To be able to successfully compensate the echos occurring in bidirectional telecommunications systems, comprising for example 2-wire/4-wire converters and/or AD/DA converters and/or satellite transmission links, the echo coupling factor and echo delay time for a system must be determined with a very high degree of accuracy and independently of the double talk situation. For this purpose, in accordance with the invention a process is provided wherein, by analyzing a dialogue, states and time segments are determined in which the probability of an incorrect evaluation of the coupling factor between the transmission paths and the echo delay time is very low. To improve the measurement accuracy, auxiliary quantities dependent upon the speech dynamics and the characteristic curve of the speech signal are introduced.

Abstract: A method of controlling an echo canceller for providing an echo cancellation signal by subtracting an echo replica signal either from a transmit signal or a receive signal. The method includes the steps of a first step of performing a speech detection based on the echo cancellation signal provided by the echo canceller; a second step of detecting a double talk condition based on a result of the speech detection performed by the first step; and a third step of controlling a learning operation of the echo canceller based on a result of the double talk detection performed by the second step.

Abstract: Residual echo components resulting from an adaptive echo canceler are minimized by employing a subband echo suppressor. The residual echo components are minimized by individually, controllably adjusting the echo suppressor attenuation level in each of the subbands to gradually attenuate echo peaks having a magnitude lower than a clipping threshold with the magnitude of the attenuation increasing for residual echoes of decreasing magnitude. Additionally, the threshold of suppression engagement is dynamically matched to the signal gain in the echo path for each subband. If the signal gain in a subband is high (strong echo), the suppressor must attenuate residual echo peaks in the subband of higher energy to maintain a constant quality of service.

Abstract: A telephone system including a base station and a mobile unit in communication with each other is provided with an acoustic echo canceler. A sound signal is input. A reference signal that matches the sound signal is formed. A plurality of encoders and decoders receive the reference signal and form a modified reference signal. The plurality of encoders and decoders match a set of encoders and decoders that operate on the sound signal in the base station and the mobile unit. An adaptive filter receives the modified reference signal and forms a replica signal. A subtractor subtracts the replica signal from an echo signal received by the base station.

Abstract: A speakerphone incorporates a portion defining a loudspeaker cavity, the cavity containing a loudspeaker. The speakerphone also includes a handset with an internal microphone in the handset. The speakerphone has a housing adapted to receive the handset, and when it is thus received, the handset microphone is acoustically coupled to the loudspeaker cavity, allowing the handset microphone signal to be used to control feed-back or feed-forward algorithms that enhance the performance of the speakerphone.

Abstract: A hands-free communications unit employs a downlink peak power memory element (150, FIG. 1), a downlink voice activity detector (140), and an uplink voice activity detector (180). The downlink voice activity detector (140) evaluates the incoming signal to determine if the signal includes voice. In a similar manner, the uplink voice activity detector (180) determines if an outgoing signal includes voice. In the event that the incoming signal includes voice and the energy of the outgoing signal exceeds a threshold calculated from the peak power of the incoming signal, the uplink voice activity detector (180) indicates that a double-talk condition exists (350, FIG. 3). In the event that the incoming signal does not include voice, the uplink voice activity detector (180) compares the uplink signal to a voice threshold above a noise floor to determine if the outgoing signal includes voice.