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: 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: 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: 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: An active noise suppression system for use in noisy environments includes a dual microphone noise suppression system in which the echo between the two microphones is substantially canceled or suppressed. Noise is cancelled by the use of first and second line echo cancellers, which model the delay and transmission characteristics of the acoustic path between the two microphones. In a first embodiment, a noise suppression system acts as an ear protector, canceling substantially all or most of the noise striking the dual microphones of the ear set. In a second embodiment, a noise suppression system in accordance with the present invention acts a noise suppression communication system, suppressing background noise while allowing speech signals to be heard by the wearer.

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: A pure echo delay estimate of an echo path impulse response estimate represented by echo canceller filter coefficients is calculated. This is done by incrementing (S9) an assumed pure delay m, where m is a positive integer representing an offset of m filter coefficients into the echo path impulse response estimate, until (S8′) a predetermined filter coefficient dependent decision function exceeds a predetermined threshold function, and by representing (S10) the pure delay by the smallest integer mSTART that makes the decision function exceed the threshold function.

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: A communication circuitry for providing effective detection of tones in a communication network. The communication circuitry comprises filtration circuitry, summation circuitry, detection circuitry, and interface circuitry. The filtration circuitry is coupled to the summation circuitry and is configured to receive a communication, process the communication to estimate an echo return in the communication, and provide the estimated echo return to the summation circuitry. The summation circuitry is coupled to the detection circuitry and configured to receive the estimated echo return from the filtration circuitry, receive a tone transmission and echo return from the interface circuitry, subtract the estimated echo return from the tone transmission to generate a substantially pure tone, and provide the substantially pure tone to the detection circuitry. The detection circuitry is configured to receive the substantially pure tone and process the substantially pure tone to perform a programmed operation.

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: An echo canceler for suppressing an echo of a receive signal contained in a transmit signal directed to a telephone circuit and transferring the transmit signal subjected to echo suppression to the telephone circuit. The echo canceler includes a pseudo echo path for attenuating and delaying the receive signal supplied from the telephone circuit and adding the delayed receive signal to the transmit signal, and an echo cancel circuit for estimating a transfer function of the pseudo echo path on the basis of the receive signal supplied from the telephone circuit and the echo of the receive signal, generating a pseudo echo on the basis of a value of the estimation and the receive signal, subtracting the pseudo echo from the transmit signal passed through the pseudo echo path, and thereby transferring the transmit signal subjected to echo suppression to the telephone circuit.

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: The invention relates to a circuit configuration for two-wire/four-wire conversion. The circuit configuration has a receiver which is coupled on an input side to a digital two-wire reception path via which its receives a digital signal. A signal-processor is connected downstream of the receiver and emits a first and second signal. An echo-canceling device receives the first signal and emits a third signal. A digital-to-analog converter is provided which receives the second signal and emits a fourth signal. A hybrid circuit is coupled to an analog four-wire transmission-reception path, receives the fourth signal, and emits a fifth signal. An analog-to-digital converter receives the fifth signal and emits a sixth signal. A transmitter receives the sixth signal and is coupled on an output side to a digital two-wire transmission path and emits a digital transmitted signal to the latter. In the invention, the digital-to-analog converter has a sigma-delta modulator.

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 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 hybrid echo suppresser and associated methods of use provide echo suppression in a bi-directional communications system, while minimizing desired-voice signal disruption. The hybrid echo suppresser includes a linear echo canceller (LEC), a non-linear processor (NLP), and a control circuit. The LEC operates on an echo-causing signal to produce an estimated-echo signal, which is subtracted from an input echo-containing signal, thereby producing an echo-cancelled signal. The NLP operates on this echo-cancelled signal to eliminate residual echo arising from system non-linearities. The control circuit repeatedly compares the echo-containing signal and the estimated-echo signal to determine if the echo-containing signal includes a desired-voice component. If not, the control circuit operates the NLP in an echo-voice mode having substantial signal attenuation.

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: The invention includes a system for processing telecommunications signaling. Signaling for a call is received into a signaling processor. The signaling processor processes the call and generates new signaling incorporating the processing. The new signaling is transferred to network elements as appropriate. The signaling processor is not coupled to a switch matrix and only communicates with the network elements over signaling links.

Abstract: An apparatus and method for estimating a time delay between two signals are provided. An original signal and a delayed signal are provided to an average magnitude difference processor for determining an average magnitude difference for a sum of samples representing an envelope of each of the respective original and delayed signals, each for a series of sample delays. The time delay between the two signals is determined as the sample delay which minimizes the average magnitude difference.

Abstract: Techniques for enhancing the sound quality of computer telephony systems are provided. In general, digital signals including telephony sounds are converted from the time domain to the frequency domain. If noise is detected in the frequency domain conversion of the digital signals, the noise is removed utilizing a filter. The noise detection and filtering can be performed in software so that enhanced audio quality can be achieved at a relatively low cost and with flexibility for very diverse environments.

Abstract: A digital signal processing system and method, such as for echo cancellation in voice processing systems, is described.

Abstract: The amplitudes of a signal on the microphone input of a telephone and a signal on the line input of the telephone are compared with each other and with two thresholds. The comparisons are sampled and the samples control an up-down counter. The count in the counter is compared with two thresholds and the results of the comparisons are used to control at least two accumulators. A first accumulator counts consecutive comparisons having the same result and a second accumulator counts up or down in accordance with the comparisons. The counts in the first and second accumulators are logically analyzed to determine whether to receive or transmit. A plurality of samples defines a window, at the end of which the up-down counter is reset. The accumulators store data from a plurality of windows.

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: An object of the present invention is to provide an echo canceller capable of generating, when the double-talk detection precision is degraded and the learning of a filter coefficient swings by noises, an echo replica signal using a filter coefficient having a low degree of error at the time of detecting a double-talk state. A backup filter coefficient is calculated by adding a predetermined weighting factor times a filter coefficient obtained by the learning and a predetermined weighting factor times the preceding backup filter coefficient stored in second storage means.

Abstract: An echo canceling apparatus suitable for multiplexed transmission lines is provided that achieves a short convergence time and a small number of arithmetic operations. Adaptive filters are located in respective transmission lines. Each adaptive filter adaptively controls tap positions such that tap coefficients are located in dispersive regions which are obtained by removing fixed delays from the impulse response in an echo path. A controller receives tap position information from the adaptive filters to mutually compare the convergence degrees among the adaptive filters. Thus, the controller respectively supplies coefficient update control signals to the adaptive filters so as to suppress coefficient adaptation of an adaptive filter whose convergence is leading compared with the others.

Abstract: A system and method for detecting and controlling severe echo. The system includes an echo canceller having an adaptive filter responsive to a set of coefficients, a first memory storing a software program operative to evaluate these coefficients to determine the severity of echo existing in the system, and a second memory accessible by the echo canceller and the first memory which stores the coefficients. Upon evaluating the coefficients, the system and method enables the proper mode of echo cancellation in the echo canceller.

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: In accordance with the invention, from the characteristic curve of the speech signal there are derived pulse sequences which are dependent upon the frequency but not the amplitude of the speech signal, so that in the case of a real echo, independently of the attenuation, identical pulse sequences occur in the case of the transmitted signal and, with a time shift, in the case of a received echo. The correspondence between the pulse sequences of transmitted signal and received signal is an indication that the received signal is an echo of the transmitted signal.

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 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 adaptive filter (and a corresponding adaptive filtering method) for use in an echo cancellation environment, comprising a finite impulse response filter having a vector of adaptive coefficients, and an adaptive coefficient vector update module which modifies the vector of adaptive coefficients in response to an error signal and based on a predetermined probability density function for said error signal. In accordance with several illustrative embodiments of the present invention, the probability density function is advantageously non-Gaussian, and has certain advantageous characteristics which provide for robustness in the adaptive algorithm based thereupon.

Abstract: In a communication system (100), a first echo return loss through communication system (100) associated with a first call connection between a caller (111) and a first caller (114) is estimated. A second echo return loss associated with a second call connection between caller (111) and a second caller (112) is estimated. The call connection may be switching from the first call connection to the second call connection after estimating the first echo return loss, and from the second call connection to the first call connection after estimating the second echo return loss. Echo generated in the first call connection is cancelled based on at least the first echo return loss by an echo canceller in communication system (100) after switching from the second call connection to the first call connection.

Abstract: An echo and noise suppression device for processing an information signal to suppress unwanted signal components. In an exemplary embodiment, the echo suppression device includes an AC-center clipper having a clipping window with an adjustable window center. A clipping threshold of the AC-center clipper is set to attenuate the unwanted signal components, and the center of the clipping window is varied in accordance with the value of the information signal so that the unwanted components are suppressed even when the information signal contains a significant noise component. In another exemplary embodiment, a mobile radio station is provided including a microphone for receiving a near-end audio input and for producing a near-end audio output which is to be transmitted to a far-end user. The exemplary mobile station also includes a loudspeaker for broadcasting, to a near-end user of the mobile station, a far-end audio signal which is generated by the far-end user and then received at the mobile station.

Abstract: Echo cancelling in a communication line system, particularly an ISDN or XDSL system, is performed by means of a tunable hybrid (5) which comprises tunable passive elements and is integrated in the analog part of the front end (3), whereby the values of the tunable passive elements are controlled by digital control means (4), for example a microprocessor.