Abstract: Disclosed is a hardware architecture receiving multi-input blind source signals and obtaining multi-output. An apparatus for separating blind source signals includes: a forward process unit receiving a plurality of blind source signal vectors and outputting a plurality of output signal vectors by using a predetermined blind source separation algorithm; an update process unit receiving the plurality of output signal vectors and learning first weighting values used for the predetermined blind source separation algorithm according to a predetermined learning algorithm; and a weight process unit having a matrix operation structure for receiving the first weighting values and converting them into coefficients and second weighting values applicable to the predetermined blind source separation algorithm. The forward process unit includes (L+1) identical processing elements connected in a systolic array structure, where L is the number of sequential delay of blind input signal vectors.

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 line card including: a co-channel estimator and a code selector. The line card is configured to couple to digital subscriber lines to support multi-tone modulation of communications channels thereon. The co-channel estimator is configured to estimate co-channel crosstalk coupling coefficients among selected pairs of the subscriber lines at levels for which the total crosstalk into a selected victim line among the plurality of digital subscriber lines substantially corresponds to the sum of the products of the corresponding crosstalk coupling coefficient for each remaining disturber one of the plurality of subscriber lines and a corresponding substantially unique vector transmitted thereon. The code selector couples to the co-channel estimator. The code selector is configured to select a cross-talk estimation code type and to generate substantially unique code vectors derived there from for injection into selected ones of the of subscriber lines.

Abstract: A method and apparatus for reducing crosstalk between digital subscriber lines includes: firstly, determining a parameter of VDSL (Very-high-bit-rate Digital Subscriber Line); secondly, determining frequency range of signal to be transmitted in the line and transmitting signal in the corresponding line according to the determined frequency range. This mainly means reducing or turning off the transmitting of the signal at a high frequency band and only using low frequency band for data transmitting in the case that the line parameter exceeds the predetermined threshold, so that crosstalk to neighboring lines is reduced. The present invention can effectively overcome the crosstalk against neighboring lines during the channel discovery phase of digital subscriber line and the transfer performance of the digital subscriber line in operation can be guaranteed. The present invention is realized based on the established standard, so the present invention is simple and easy to implement.

Abstract: An echo detector has a correlator for correlating between an incoming signal and an echo signal, the correlator being arranged to operate recursively, the detector being arranged to detect the echo and the length of its delay from peaks in the correlation. Recursive operation means the computational load can drop from being proportional to N*N, to being proportional to 2N, where N is a number of samples in a correlation window. It also reduces computation delay. A filter at the inputs of the correlator suppresses low frequencies corresponding to the strongest components of pitch in human speech. Such filtering can remove periodic components which can mask a distinct peak of an echo. A classifier thresholds a ratio of highest and next highest correlation peaks beyond a time margin around the highest correlation peak, to reduce erroneous detections. It can classify two or more ratios simultaneously as distinct echoes.

Abstract: A telephone system comprising an analog telephone and a full-duplex speakerphone adapter is disclosed. The analog telephone comprises an ordinary analog telephone subscriber circuit for transmitting and receiving analog signals and a handset for users. The full-duplex speakerphone adapter is coupled between the analog telephone and a central office, uses a subscriber loop interface circuit through the first telephone line to couple to the ordinary analog telephone subscriber circuit of the analog telephone and uses a telephone hybrid interface circuit through the second telephone line to couple to a wall jack to communicate with the central office.

Abstract: A telephony system equipped with an echo cancellation module is disclosed. A hybrid interface circuit outputs an outbound signal to a wall jack while receiving an inbound signal from the wall jack. The inbound signal may contain a line echo of the outbound signal caused by the impedance mismatches in the hybrid interface circuit. A line echo canceller containing an adaptive filter is used to cancel the line echo in the inbound signal based on the learned line echo path characteristics captured in the training period. During a brief period after the telephony system is activated, the line echo canceller enters into a calibration mode wherein its adaptive filter is trained to learn the line echo path characteristics. During the calibration mode, the line echo canceller generates a calibration signal as the outbound signal and receives the line echo from the hybrid interface circuit to perform learning of the line echo path characteristics.

Abstract: Plural echo cancellors each are tuned for different hybrid delays until it is established which echo cancellors are operating to cancel echos. The echo cancellors not so operating are then terminated.

Abstract: A packet voice transceiver adapted to reside at a first end of a communication network and to send and receive voice packets to and from a second packet voice transceiver residing at a second end of the communication network. The transceiver includes a voice encoder, a packet transmitter, a packet receiver, a voice decoder and a far-end echo canceller. The voice encoder receives an ingress audio signal and encodes the audio signal to produce an ingress packet voice signal. The packet transmitter transmits the ingress packet voice signal over the network to the second packet voice transceiver. The packet receiver receives an egress packet voice signal transmitted over the network by the second packet voice transceiver. The voice decoder decodes the egress packet voice signal to produce an egress audio signal. The far-end echo canceller reduces echo that is present in the egress audio signal and that originated at the second end.

Abstract: A method and system in network switch for cancelling multiple sources of network echo, including echo that is combined with a persistent, non-echo signal. A first echo canceller cancels a first echo signal that arises from an echo path in the network switch, and second echo canceller cancels a second echo signal that arises from an echo path in an external network device. The first echo canceller operates by first removing any persistent, non-echo signal that may be combined with the first echo signal, then training on the remaining, pure echo signal. The second echo canceller includes two sub-component echo cancellers that are coordinated so as to provide a minimum error signal for cancellation applied to the second echo signal.

Abstract: A nonlinear echo compensator compensates for nonlinear echo in a transceiver including a transmitter line driver with current cells that are operated in a Class B operating mode. A mapping circuit generates a pattern dependent driving signal. A coefficient generator generates a compensator coefficient. A canceling circuit communicates with the mapping circuit and the coefficient generator and compensates for nonlinear echo in a received signal based on the pattern dependent driving signal and the compensator coefficient.

Abstract: A method of communicating data across a channel that experiences near-end cross talk (NEXT) interference and far-end cross talk (FEXT) interference in alternate intervals. In one embodiment, the method comprises: a) determining NF, the number of bits per symbol usable in a FEXT-only mode of operation; b) determining NS, a number of bits per symbol usable in a single mode of operation; c) determining whether the FEXT-only mode or the single mode provides a higher data rate; and d) configuring a modem to transmit using the mode having a higher data rate. The FEXT-only mode may be determined to have a higher data rate when 126NF>340NS.

Abstract: An echo canceller circuit is set forth. The echo canceller circuit includes a digital filter having adaptive tap coefficients to simulate an echo response occurring during a call. The adaptive tap coefficients are updated during the call using a Means Squares process. A tap energy detector is also employed. The tap energy detector identifies and divides groups of taps having high energy from groups of taps having low energy. The high energy tap groups are smaller in number than the low energy tap groups. The high energy tap groups are adapted separately from the low energy tap groups using the Least Squares process. Still further, the high energy tap groups may be adapted using an adaptive gain constant ? while the low energy tap groups are adapted using an adaptive gain constant ??, wherein ?>??.

Abstract: A method and system of multi-channel echo cancellation using round robin regularization. The multi-channel round robin regularization echo cancellation method includes applying a plurality of adaptive filters, each having an inverse correlation matrix, to the multi-channel playback signal. Each of the plurality of adaptive filters is selected in a round robin sequence, so that every round each of the filters is selected. The inverse correlation matrix associated with each selected adaptive filter then is regularized as needed. The regularized adaptive filter then is used to remove the echo of the multi-channel playback signal from a captured signal. Regularization is implemented in a round robin manner to ensure that each subband is selected so that the adaptive filter for that subband can be examined. Other features of the multi-channel echo cancellation system and method include dynamic switching between monaural and multi-channel echo cancellation and mixed processing for lower and upper subbands.

Abstract: An echo canceller used in a vocal information exchange system operates in such a circumstance that a sound wave released by a speaker is picked up by a microphone unintentionally and works to suppress an echo signal included in the audio signal produced by the microphone by using a pseudo echo signal which is produced based on the incoming audio signal to be fed to the speaker and produce a subtracted audio signal which becomes the outgoing audio signal, with the echo signal being suppressed based on the accurately-calculated degree of echo suppression achieved by use of the pseudo echo signal.

Abstract: A method for reducing transmit echo in a DSL modem comprises selecting at least one cancellation device of a plurality of cancellation devices. An attenuation signal is generated using the selected cancellation device. At least a portion of transmit echo is removed from a receive signal using the attenuation signal.

Abstract: According to one exemplary embodiment, a communication device comprises a line driver, where the line driver provides an output signal. The communication device further comprises a digital echo canceller module coupled to the line driver, where the digital echo canceller module receives an echo reference signal determined by the output signal of the line driver, where the echo canceller module outputs an echo cancellation signal, and where the echo cancellation signal is capable of canceling linear and non-linear components of a composite signal. The communication device further comprises a summation module coupled to the digital echo canceller module, where the summation module receives the echo cancellation signal and the composite signal, and where the composite signal comprises a received signal and the linear and non-linear echo components. The summation module is configured to subtract the echo cancellation signal from the composite signal and to output the received signal.

Abstract: An audio communications system has an acoustic echo cancellation (AEC) module. The AEC module receives a digital signal sent to a loudspeaker and a digital signal received from a microphone. The signal received from the microphone contains an echo of the signal played through the loudspeaker. The loudspeaker signal is processed by an audio generation module (AGM) that models substantially nonlinear distortions that can occur while producing the signal played through the loudspeaker. The AGM includes a modeling path comprised of one or more distortion modules. Each distortion module receives digital samples as input, modifies the samples to model a form of distortion, and outputs the modified samples. The output of the AGM is provided to an acoustic echo estimation (AEE) module that uses adaptive algorithms to compensate for substantially linear changes in the echo characteristics of the environment in which the loudspeaker and microphone are located.

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: In a system in which echo transferred from a received signal to an outgoing signal is canceled by subtracting an adaptively predicted echo signal from the outgoing signal, howling is detected by detecting the effect of the predicted echo signal. The detected effect preferably depends on the magnitude or polarity of the predicted echo signal. The effect may be detected by comparing an envelope of the received signal or outgoing signal with a corresponding envelope of the residual outgoing signal left after echo cancellation. The echo attenuation on the echo path from the received signal to the outgoing signal may also be calculated, the conditions for howling detection being made more stringent as the detected echo attenuation increases. When howling is detected, it may be suppressed by attenuating the residual outgoing signal or initializing an adaptive filter that generates the predicted echo signal.

Abstract: A communication system having an echo canceller is disclosed. One embodiment of the echo canceller includes an adaptive filter used to provide an estimate of reflected echo which is removed from the send signal. The echo canceller may also include a near-end talker signal detector which may be used to prevent the adaptive filter from adapting when a near-end talker signal is present. The echo canceller may also include a nonlinear processor used to further reduce any residual echo and to preserve background noise. The echo canceller may also include a monitor and control unit which may be used to monitor the filter coefficients and gain of the adaptive filter to maintain stability of the echo canceller, estimate pure delay, detect a tone, and inject a training signal. The echo canceller may also include a nonadaptive filter used to reduce the length of the adaptive filter.

Abstract: An echo canceller includes: an echo cancellation part canceling echo included in a near-end input signal; a noise suppression part employing a low-pass filter having a variable cut-off frequency adaptive to power of an input signal, the noise suppression part removing a residual echo component from an output of the echo canceller part; a threshold generation part generating a variable threshold varying in accordance with a power level of a far-end input signal; a determination part making a determination based on a comparison between a power level of the output of the echo cancellation part and the variable threshold; and a processing part performing noise suppression using the low-pass filter on the output of the echo cancellation part when the determination part determines that the power level of the output of the echo cancellation part is lower than or equal to the variable threshold.

Abstract: Providing multi-path echo cancellation comprises providing a main echo signal corresponding to a main echo section of an echo response and providing a plurality of tail echo signals, each of the plurality of tail echo signals corresponding to respective tail echo sections of the echo response. Next, providing a multi-path echo canceller comprises summing the main echo signal and the plurality of tail echo signals to provide a synthesized echo.

Abstract: An echo canceller circuit for use in an echo canceller system is set forth that provides sensitive double-talk detection. The echo canceller circuit comprises a second digital filter having adaptive tap coefficients to simulate an echo response occurring during the call. The adaptive tap coefficients of the second digital filter are updated over the duration of the call using a Least Mean Squares process having an adaptive gain a. A channel condition detector is used to detect channel conditions during the call. The channel condition detector is responsive to detected channel conditions for changing the adaptive gain a during the call. For example, the channel condition detector may detect the presence of a double-talk condition and set the adaptive gain a to zero. Similarly, the channel condition detector may detect the occurrence of a high background noise condition and set the adaptive gain a to a level less than 1 that is dependent on the detected level of the background noise.

Abstract: An apparatus and method for echo cancellation that detects and tracks the active regions of the echo channel impulse response rather than the entire echo channel impulse response and uses this sparse model to cancel the echo component of the received signal. Two sets of filter coefficients are used, one to remove and track the active regions and one to detect the active regions.

Abstract: A speech recognition system includes a user profile to store acoustic data and a corresponding text transcript. A speech recognition (“SR”) server downloads the acoustic data and the corresponding text transcript that are stored in the user profile. A speech recognition engine is included to adapt an acoustic model based on the acoustic data.

Abstract: Techniques to cancel echo utilizing reduced die space and reduced usage of a time shared processor.

Abstract: A noise level calculator for detecting noise in a telephone line echo canceller that utilizes adaptive filters enables silencing of noise switching so that it is not heard on the far end side. The calculator promotes determining whether the reference signal applied to the adaptive filter is a noise or non-noise segment. The calculator operates by calculating the difference in energy of two sample windows, calculating the difference in the energy of the signal within each of the sample windows, updating a variance parameter based on the difference, and performing further alternative operations based on the value of the variance parameter.

Abstract: This invention describes a mobile phone with a means of measuring a background sound level of an environment. When a user either initiates or receives a phone call, sound levels before and during the call are compared. Once it is decided that the voice is too loud based on a predetermined criteria, the phone gives the user a feedback indicating that the voice is too loud and potentially a disruption to other people. Furthermore, the present invention provides a feedback for a voice adjustment by utilizing a sidetone adaptive signal that filters the user own speech directly to an earpiece.

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: A downlink activity and double talk probability detector and method for an echo canceler circuit (10) improves the stability of an echo canceler adaptive filter (300) and improves the attenuation of post-echo canceler uplink data (388). The echo canceler circuit (10) includes a downlink activity and double talk probability data generator (30) and an echo canceler stage (20). The downlink activity and double talk probability data generator (30) receives pre-echo canceler uplink data (40) and downlink data (50) and in response produces double talk probability data (60) and downlink activity data (70). The echo canceler stage (20) receives the downlink data (50), the pre-echo canceler uplink data (40), the double talk probability data (60) and the downlink activity data (70), and in response produces uplink data (80).

Abstract: A distributed echo cancelling architecture is provided where echo-cancelling functions are performed at locations remote from devices receiving signals with echoes. The echo cancelling functions use a reference signal, which has been corrupted with the echoes at the devices, for echo cancellation. As echo canceller resources are located at a central system and not at each individual device, the echo canceller resources can therefore be shared between the devices.

Abstract: Method and apparatus for signal reflection removal, such as echo cancellation, is described. Signal samples are delayed. The delays allow burst processing of consecutive samples of transmitting and receiving signals in a communication network, such as in a telephone communication system. As a result, there is tremendous reduction of memory bandwidth when compared to conventional sample-by-sample processing of signals. Echo cancellers described herein can be implemented in an FPGA. Echo in over a thousand channels can be cancelled using an FPGA and an external memory device. In embodiments for reflected signal cancellation, multiple stages of echo estimation are used.

Abstract: An optimal echo canceling processing is provided regardless of a condition of a partner's system and even in the condition with large delay amount and large fluctuation amplitude of a network transmission. A speaker 10 inputs a voice via a microphone 11. A voice signal of this voice is transmitted to a terminal 14 of a conversation partner via a VoIP application 13 and the internet 30. Concurrently, the voice signal is inputted to an echo canceller 100. The echo canceller 100 detects sound characteristics of an echo path in advance or dynamically, adjusts a filter coefficient for generating an echo canceling signal, and receives an adjustment by a user. The echo canceller 100 generates the echo canceling signal by processing the received voice signal based on the sound characteristics and the adjusting amount. The echo canceling signal is subtracted from a response signal containing an echo that has been re-inputted from a loudspeaker 22 of the partner via a microphone 21.

Abstract: The invention relates to a communications network, including switching centers having no echo canceling equipment and switching centers having echo canceling equipment. The switching centers are capable of changing the call party or the call during the call. The switching center which controls the change of call party or call type is arranged to command the nearest switching center having the echo canceling capability to connect/disconnect an echo canceling equipment to/from the connection. In other words, in these special situations the echo cancellation is controlled by a new signaling feature within the communications network.

Abstract: Briefly, in accordance with one embodiment of the invention, a method for actively characterizing the latency of an audio channel of a computer, such as a personal computer, is provided. At least two signal streams for a waveform are created in the audio channel. The presence of the first signal sample stream for the waveform and the second signal sample stream for the waveform is detected at a point in the audio channel. The time between the detections of the signal sample streams is measured. Briefly, in accordance with another embodiment of the invention, a method of actively characterizing the latency of an audio channel of a computer, such as a personal computer, is provided. At least a first and a second waveform are created in the audio channel. The presence of the first and the second waveform are detected at a point in the audio channel. The time between the detections of the waveforms is measured.

Abstract: A double talk activity detector (30) and method for an echo canceler circuit (10) improves the probability of detecting a double talk condition based on at least pre-echo canceler uplink data (40). The echo canceler circuit (10) includes a double talk activity probability data generator (30) and an echo canceler stage (20). The double talk activity probability data generator (30) receives pre-echo canceler uplink data (40) and in response produces double talk activity probability data (50). The echo canceler stage (20) is coupled to the double talk activity probability data generator (30) and receives downlink data (60), pre-echo canceler uplink data (40) and the double talk activity probability data (50) and in response produces attenuated uplink data (70).

Abstract: A system for updating filter coefficients for use with an echo canceller, a related method for updating filter coefficients and an echo canceller including the system or the method. In one embodiment, the echo canceller has a nonadaptive filter and an adaptive filter, the nonadaptive filter having nonadaptive filter coefficients and configured to develop a nonadaptive error signal as a function of the nonadaptive filter coefficients, and the adaptive filter having adaptive filter coefficients and configured to develop an adaptive error signal as a function of the adaptive filter coefficients. The system includes a monitoring subsystem configured to monitor a quality of the nonadaptive and adaptive error signals and a selection subsystem configured to select one of the nonadaptive and adaptive filter coefficients for use by the nonadaptive filter based on the quality of the nonadaptive and adaptive error signals.

Abstract: A wraparound delay amount detecting part calculates a cross-correlation r(k) from an output speech signal “ai” supplied to a loudspeaker and an input speech sigal “bi” inputted through a microphone array to obtain a delay amount “d” of a wraparound speech signal. The delay processing part generates a speech signal “ai-d” obtained by delaying the output speech signal “ai” by the delay amount “d”. Even if there is a change in delay amount due to the variation in environment, appropriate delay processing can be conducted by the delay processing part. In an adaptive filter, an estimated wraparound speech signal ai-d? is generate from the speech signal “ai-d” subject to delay processing. A subtracter subtracts the estimated wraparound speech signal ai-d? from the input speech signal “bi” to generate an echo cancellation signal “ei”. A coefficient updating part updates the coefficient of the adaptive filter.

Abstract: An adaptive filter is programmed with an algorithm based on a normalized Least Mean Squares (nLMS) algorithm that adapts each sample time. The algorithm is modified to be more efficient in a variety of DSPs by computing multiple errors, one per sample, before updating coefficients. The update equation utilizes the multiple errors to achieve adaptation at a similar performance to known nLMS algorithms that adapt each sample time but without the instability that is observed in low echo-to-near-end-noise ratio (ENR) input conditions. Varying the relaxation step size prevents divergence. The DSP utilizes either one or more MAC units.

Abstract: A telephonic handset comprises an active noise reduction (ANR) system. The ANR system comprises a reference microphone and an IIR filter. The IIR filter is receivingly coupled to the reference microphone with respect to noise reference signals, and it is transmittingly coupled to the receiver transducing element of the handset. The ANR system is configured as a fixed feed-forward noise cancellation system. Preferably, the IIR filter has a transfer function derived, in part, from the open-loop gain of a feedback noise cancellation system. In specific embodiments of the invention, the noise reference microphone is situated so as to sample the ambient noise field near the front face of the receiver, but without directly sampling the noise field on the front face.

Abstract: A terminal in a communication system includes emitting and receiving transducers and an echo processor having an adaptive echo canceler that processes a residual echo output from the adaptive echo canceler.

Abstract: The present invention provides for adaptive filters that have improved computational and memory bandwidth proprieties. When applied to telecommunication applications, the present invention additionally provides for improved methods and systems of canceling echoes. In one embodiment of the adaptive filter of the present invention, a filter, preferably an adaptive finite impulse response (FIR) filter, of an appropriate length, N, is chosen. Once the filter is chosen, convergence is achieved and the filter is converted to an infinite impulse response (IIR) filter. In the course of operation, data is received from an input source and used to adapt the zeroes of the IIR filter using the least means square (LMS) approach, keeping the poles fixed. The adaptation process generates a set of converged filter coefficients that are then applied to the input signal to create a modified signal used to filter the data.

Abstract: A system for performing echo cancellation via use of a transceiver interface circuit. In a simplified embodiment, a digital signal processor (DSP), an analog front end (AFE), a line driver, and a hybrid network are located within the circuit. The DSP is configured to process a transmit, receive, and a transmit error signal. Mathematical operations of the DSP eliminate error from the receive signal with the help of the transmit error signal. The AFE modifies the transmit, receive, and transmit error signal from analog to digital or vice-versa. The line driver amplifies the transmit signal to a power level compatible with a transmission network. The line driver further produces the transmit error signal which is created by the amplification of the transmit signal. The hybrid network receives the amplified transmit signal from the AFE and a receive signal from the transmission network, and isolates the transmit signal from the receive signal.

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 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: Methods and apparatus for processing a received voice signal from a variable-delay network by a communications system that has at least one echo canceller. An echo canceller determines delay characteristics that are associated with the received voice signal and correspondingly modifies echo cancellation processing. In a variation of the embodiment of the invention, a first echo canceller sends silence descriptor information to a second echo canceller. The second echo canceller reconstructs a voice signal based upon the silence descriptor information and excitation vector information.

Abstract: The present invention provides an acoustic echo canceler for use in a telephone set having a handsfree conversation function. This acoustic echo canceler simultaneously achieves both the stabilization and speedup of an estimation operation and also attains the more improvement of speech quality. Moreover, the present invention provides a handsfree telephone set employing this acoustic echo canceler. Furthermore, this acoustic echo canceler is operative to cancel an echo generated by inputting an output of an acoustic device to itself substantially simultaneously with the outputting the output thereof. Further, this acoustic echo canceler comprises an adaptive filter for performing an operation of estimating the frequency characteristic and the phase characteristic of a part, whose frequency characteristic and phase characteristic vary, on an acoustic path and a fixed coefficient filter for simulating a part, whose frequency characteristic and phase characteristic are fixed, on the acoustic path.

Abstract: A modem incorporating apparatus and methods to achieve computationally efficient echo cancellation. The apparatus include a cyclic echo synthesizer sub-canceler in the time domain and a echo canceler in the frequency domain. The method includes generating a cyclic echo synthesizer signal using a sub-canceler structure, adding the cyclic echo synthesizer signal to a receive signal in the time domain, generating an echo cancellation signal, and subtracting the echo cancellation signal from the receive signal in the frequency domain. The apparatus and methods may be used for echo cancellation in an asynchronous digital subscriber line (ADSL) modem using discrete multi-tone (DMT) technology.

Abstract: A method for suppressing echo signals generated in a communication path such as acoustic coupling between a speaker and a microphone in a telephone device handset includes the steps of monitoring signals supplied to the communication path to determine an attribute thereof and masking signals received from the communication path as a function of the determined attribute of the monitored signals thereby to suppress echo.