Abstract: A method of masking a residual echo signal by an echo canceller is provided. The method comprises receiving a far-end signal, adjusting filter coefficients of an adaptive filter in response to the far-end signal, generating an echo model signal based on the far-end signal using the adaptive filter, receiving a near-end signal, subtracting the echo model signal from the near-end signal to generate an output signal, defining a spectral mask based on the near-end signal, wherein the spectral mask is indicative of near-end spectral peaks and near-end spectral valleys, de-emphasizing the output signal in spectral regions of the near-end spectral peaks, and emphasizing the output signal in spectral regions of the near-end spectral valleys, wherein the de-emphasizing occurs during filter coefficients determination for the adaptive filter. A weighted filter may perform the de-emphasizing and the emphasizing operations, where the weighted filter uses medium term spectral characteristics of the near-end signal.

Abstract: A broadside small array microphone beamforming apparatus comprises first and second omni-directional microphones, a microphone calibration unit, and a directional microphone forming unit. The first and second omni-directional microphones respectively convert voice from a desired near-end talker into first and second signals. The second and first omni-directional microphones and the desired near-end talker are respectively arranged at three points of a triangle. The microphone calibration unit receives the first and second signals and correspondingly outputs first and second calibration signals. The directional microphone forming unit receives the first and second calibration signals to generate a first directional microphone signal with a bidirectional polar pattern. The adaptive channel decoupling unit receives the first calibration signal and the first directional microphone signal to generate a first main channel signal and a first reference channel signal for noise detection.

Abstract: Bidirectional differential point to point simultaneous high speed signalling is provided between integrated circuits with highly effective echo canceling. Each integrated circuit comprises a transmitter for transmitting a first signal to another integrated circuit and a receiver for receiving a second signal from the other integrated circuit. The transmitter has an output buffer; a receiver has a receiver buffer and is co-located on the same integrated circuit; and a differential buffer is coupled between the input of the transmitter buffer and the output of the receiver buffer. To increase the quality of receiving the second signal, a third signal adjusted in phase and amplitude is coupled at the output of the receive buffer, so that the echoing of the first signal is canceled. Preferably, the rise time of the third signal is also adjusted.

Abstract: Techniques for suppressing large echo due to nonlinearity, instable adaptive filter, and so on. An echo cancellation system includes an echo canceller unit, a nonlinear echo detection unit, and a nonlinear echo suppression unit. The echo canceller unit (e.g., an adaptive filter) receives a reference signal and a near-end signal, derives an echo estimate signal based on the reference and near-end signals, cancels a portion of echo in the near-end signal based on the echo estimate signal, and provides an intermediate signal having residual echo. The nonlinear echo detection unit detects for large nonlinear echo in the echo estimate signal. The nonlinear echo suppression unit suppresses large nonlinear echo in the first intermediate signal based on a variable gain, which may be adjusted lower if large nonlinear echo is detected and higher otherwise.

Abstract: The invention concerns a method and device for processing echo between at least two communication devices in order to attenuate, in a picked-up signal of a communication device comprising at least one microphone, the components of the signal broadcasted by at least one other communication device comprising at least one loudspeaker, characterised in that the echo processing device comprises: means for receiving, by means of a connection with at least one other device, information representing at least one broadcasted signal of at least one other communication device, means for modifying the picked-up signal of the communication device according to information representing the broadcasted signal and information representing the coupling separating a loudspeaker of the said at least one other communication device from the microphone of the communication device.

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

Abstract: In a data processing system, a method includes receiving samples of a send path input signal (Sin) and samples of a receive path input signal (Rin); in response to receiving each sample of the send path input signal, providing a double-talk indicator to indicate whether or not double-talk has been detected; using the double-talk indicator to determine a double-talk density within a predetermined number of samples of the send path input signal or the receive path input signal; and using the double-talk density to provide a heavy double-talk indicator which indicates whether or not heavy double-talk has been detected. In response to the heavy double-talk indicator indicating that heavy double-talk has been detected, a heavy double-talk mode may be entered.

Abstract: In one aspect of the present invention, a method to reduce noise in a noisy speech signal is disclosed The method comprises: applying at least two versions of the noisy speech signal to a first filter, whereby that first filter outputs a speech reference signal and at least one noise reference signal, applying a filtering operation to each of the at least one noise reference signals, and subtracting from the speech reference signal each of the filtered noise reference signals, wherein the filtering operation is performed with filters having filter coefficients determined by taking into account speech leakage contributions in the at least one noise reference signal.

Abstract: Hands-free phones with voice activity detection using a comparison of frame power estimate with an adaptive frame noise power estimate, automatic gain control with fast adaptation and minimal speech distortion, echo cancellation updated in the frequency domain with stepsize optimization and smoothed spectral whitening, and echo suppression with adaptive talking-state transitions.

Abstract: Sound signals captured by a microphone are adjusted to provide improved sound quality. More particularly, an Acoustic Echo Reduction system which performs a first stage of echo reduction (e.g., acoustic echo cancellation) on a received signal is configured to perform a second stage of echo reduction (e.g., acoustic echo suppression) by segmenting the received signal into a plurality of frequency bins respectively comprised within a number of frames (e.g., 0.3 s to 0.5 s sound signal segments) for a given block. Data comprised within respective frequency bins is modeled according to a probability density function (e.g., Gaussian distribution). The probability of whether respective frequency bins comprise predominantly near-end signal or predominantly residual echo is calculated.

Abstract: A system and method for enhancing communications through a phone. A voice communication is received from a user of the phone. A secondary signal is received from an environment in proximity to the phone. The secondary signal is processed to determine an inverse signal in response to receiving the secondary signal. The inverse signal is combined with the voice communication and the secondary signal to destructively interfere with the secondary signal for allowing a receiving party to more effectively communicate with the user.

Abstract: Techniques for simultaneously and efficiently eliminating echo and noise mixed in voice are disclosed. According to one aspect of the present invention, an adaptive filter system for eliminating noises and echo mixed in voice signal comprises a first signal adaptive filter, a first subtractor, a second adaptive filter and a second subtractor. The first signal adaptive filter estimates the noise mixed in the voice signal according to a reference noise. The first subtractor subtracts the noise estimated by the first adaptive filter from the voice signal to obtain a voice signal without the mixed noise. The second adaptive filter estimates the echo mixed in the voice signal according to a remote voice signal. The second subtractor subtracts the echo estimated by the second adaptive filter from the voice signal without mixed noise to obtain a clean voice signal.

Abstract: A state deciding unit 2 makes a decision of double talk when both received signal and transmission signal are in a speech burst, and when the amount of reduction in an echo removal level per prescribed time period exceeds a predetermined threshold, the echo removal level being represented as a level difference between the transmission signal and a residual signal passing through the echo elimination by an adaptive filter 1, and when the amount of changes of the signal level of the received signal is less than a prescribed reference value.

Abstract: A voice recognition system includes an adaptive filter and a subtractor. The adaptive filter generates a simulated talk-back voice y(n) by setting a filter coefficient simulating a transfer system in which an input voice corresponding to a voice command and a talk-back voice output from a speaker are input into a microphone and by filtering a talk-back voice x(n). The subtractor extracts the input voice by subtracting the simulated talk-back voice y(n) from mixed sound input into the microphone. With this configuration, the talk-back voice is attenuated from the mixed sound including the input voice and the talk-back voice input tedinto the microphone, and then, the mixed sound is supplied to a voice recognition engine. Accordingly, the user can input his/her voice during a talk-back operation without the need to interrupt it by pressing a speech button every time the user wishes to input the voice. The voice recognition operation time can be thus reduced.

Abstract: An input sound processor compares power at each frequency component of an input sound with a reference value, and sets multiplication points indicating frequency components at which the total power of the input sound is to be determined. A product-sum operation is performed at the multiplication points on the power at each frequency component and the square amplitude of each filter coefficient indicating the transfer characteristic from a loudspeaker to a microphone to estimate the total power of the input sound at the position of the microphone.

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

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

Abstract: A non-linear processor for use in an echo canceller is set forth. The non-linear processor includes a center clipping digital filter receiving an echo compensated signal. The non-linear processor provides a center clipped output signal having non-linear thresholds at values of +TNLP and ?TNLP. The value of TNLP is dynamically dependent, at least in part, on echo return loss measurements. To limit the processor's susceptibility to corruption from double-talk conditions, the non-linear processor inhibits the dynamic setting of the TNLP value when a double-talk condition is present. Additionally, or in the alternative, the non-linear processor locks the value of the echo return loss measurement after a predetermined number of consecutive echo return loss measurements have values falling within a predetermined range of one another. Such locking further reduces the susceptibility of the non-linear processor to corruption from double-talk conditions.

Abstract: A communications device is presented for providing bi-directional audio communications between a near-end user and a far-end user via a bidirectional communications channel. The communications device includes an adaptive echo canceller receiving a near-end audio signal and a far-end audio signal and providing an echo-canceled near-end audio signal for transmission to the far-end user via the communications channel. The adaptive echo canceller includes a first bank of analysis filters for filtering the near-end audio signal, a second bank of analysis filters for filtering the far-end audio signal, and a bank of synthesis filters for filtering sub-band echo-canceled signals generated within the adaptive echo canceller. The first and second filter banks have a frequency response optimized to reduce echo residual gain.

Abstract: A method and system for clear signal capture comprehend several individual aspects that address specific problems in improved ways. In addition, the method and system also comprehend a hands-free implementation that is a practical solution to a very complex problem. Individual aspects comprehended related to echo and noise reduction, and divergence control.

Abstract: In an on-board hands-free communication apparatus, a gain of an amplifier of a speaker for outputting an incoming voice signal from a telephone device varies with a volume set value of a speaker volume switch adjusted by an user. When the gain of the amplifier increases equal to or greater than a threshold value, a feature extraction amplification circuit performs a feature extraction amplification processing to apparently amplify output power of the incoming voice signal, and a control circuit fixes the gain of the amplifier to the threshold value regardless of the volume set value.

Abstract: An echo removing apparatus for reducing the echo caused by sound returning from a speaker to a microphone of a small-sized communication terminal, such as a portable telephone. The echo removing apparatus includes a filter unit for outputting a pseudo echo signal estimating an echo component returning to the microphone based upon a generated sound signal supplied to the speaker, a subtractor for subtracting the pseudo echo signal supplied from the filter unit from the sound generated collection signal supplied from the microphone and a first characteristics conversion unit for converting frequency characteristics of sound generated collection signals from the microphone on the frequency axis. The characteristics of the filter unit are controlled for minimizing an error component between an output signal of the first characteristics conversion unit and an output signal of the filter unit.

Abstract: A method and apparatus for reducing crosstalk in a multi-channel communication system is disclosed. In one embodiment, outgoing signals in a multi-channel environment are manipulated into a transform domain, such as the frequency domain. Thereafter, the signals may be combined and modified based on a weighting variable to create a cancellation signal. Combined processing greatly reduces system complexity and increases processing speed. After processing in the transform domain, the cancellation signal undergoes further processing to return the cancellation signal into the time domain. The cancellation signal may then be combined with received signals to cancel crosstalk or echo. A method and apparatus for crosstalk cancellation in the analog domain and digital domain are also disclosed.

Abstract: An echo cancellation circuit includes delay units at its input unit and output unit in order to adjust the delay caused by a FIFO. The delay times of the delay units are trained using a training signal, such the REVERB signal, that is a system initializing signal. In addition, an upstream pilot tone is used to tune the delay time. The performance of the echo cancellation circuit is optimized according to the device and the method described.

Abstract: A receiver for receiving an incoming signal over a communication medium includes an echo canceller, which is adapted to receive an outgoing signal transmitted over the communication medium, and to process the outgoing signal using a set of variable processing coefficients in order to generate an echo cancellation signal. A summer combines the incoming signal with the echo cancellation signal so as to generate an echo-cancelled signal. An equalizer applies an equalization operation to the echo-cancelled signal so as to generate an equalized signal. A residual echo cancellation circuit processes the equalized signal so as to adaptively update the variable processing coefficients of the echo canceller.

Abstract: A system and method for estimating a channel spectrum. The method includes: (a) receiving an input signal from a channel, where the input signal includes one or more major echoes and zero or more minor echoes introduced by the channel; (b) identifying the one or more major echoes present in the input signal; (c) identifying the minor echoes from a filtered autocorrelation function of the input signal in response to a determination that there is only one major echo; (d) identifying the minor echoes from a filtered power spectrum of the input signal in response to a determination that there is more than one major echo; (e) computing a channel spectrum estimate from the major echoes and minor echoes; where the channel spectrum estimate is usable to remove at least a portion of the one or more major echoes and one or more minor echoes from the input signal.

Abstract: A duplexer for a communication device having a transmission unit, a reception unit and a shared antenna is disclosed. The duplexer comprises a first signal path between the transmission unit and the reception unit, the first signal path comprises a filter unit filtering the reception signal, the filtering unit provides a filtered signal to the reception unit. The duplexer further comprises a second signal path between the transmission unit and the reception unit. The second signal path comprises a cancellation unit which receives a sample of the transmission signal and produces a compensation signal. The injection of the compensation signal to the first signal path reduces the leakage signal, thereby producing a substantially leakage-free reception signal.

Abstract: The present invention relates to a suppression device for an arrangement for the transmission of audio signals, in particular uttered speech, having: an echo reduction unit (4), which is arranged between an input channel (1) for receiving an input audio signal (A1) coming from a remote end and an output channel (5) for outputting an output audio signal (A2), for suppressing an echo signal contained in the output audio signal (A2), a speech activity detection unit (7) for detecting a speech signal contained in the input audio signal (A1), and, a control unit (6) for setting an echo suppression factor(s) of the echo reduction unit (4) for echo suppression.

Abstract: An active acoustic noise reduction system which comprises a single input transducer and an output actuator that are physically located next to each other in the same location. In one embodiment, the input transducer and the output actuator are a hybrid represented by a single element. The active noise reduction system is located as close as possible to the noise source and functions to generate an antinoise cancellation sound wave with minimum delay and opposite phase with respect to the noise source. The noise reduction system also comprises a non linearity correction circuit, a delayed cancellation circuit and variable gain amplifier. The system provides user control of the quiet zones generated by the system by varying the gain of the variable gain amplifier. The system provides a user with the ability In one embodiment, an echo canceler is utilized to remove echoes fed back from the output actuator.

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 system and method for providing an apparent echo return loss is provided. A communications network may include a gateway coupled to a local exchange network. The gateway may receive signals from a packet network and output signals to a circuit switched network, which may output return signals to the gateway. The local exchange network may provide an echo return loss on signals received from the gateway. The gateway may also provide an initial echo return loss to signals prior to sending the signals to the network. The echo return loss provided by both the gateway and the local exchange network may enable the gateway to determine when to trigger adaptation in order to provide estimates of echo signals generated by the network, and to subsequently eliminate these echo signals from the return signals.

Abstract: A hybrid circuit within a full-duplex transceiver transmits an outgoing signal outward on a communication channel at the same time it receives an incoming signal arriving via the communication channel, and the outgoing and incoming signals sum to form a combined signal. The hybrid circuit generates both the outgoing signal and a replica of the outgoing signal in response to an input signal, and then subtracts the replica from the combined signal in producing a received signal. The received signal includes a component derived from the incoming signal and a residual echo component having peaks resulting from a phase difference between the outgoing signal and its replica. The transceiver adjustably delays the input signal so that the residual echo component peaks occur at times other then when the received signal is being digitized, thereby minimizing the influence of the echo component peaks on the data sequence.

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 method of improving convergence of an echo canceller in a full duplex speakerphone, wherein the echo canceller includes LEC (Line Echo Canceller) and AEC (Acoustic Echo Canceller) portions, comprising the steps of capturing coefficients during operation, storing the coefficients, and utilizing the stored coefficients as default values during start-up of a subsequent call. According to the method of the present invention, the amount of echo cancellation necessary to provoke a save of the coefficients decreases with time when it is not achieved by the system despite the presence of a reference signal (i.e. speakerphone signal).

Abstract: A system and method for signal analysis in a network. The method includes attempting, by a first processor, to compute optimal coefficients for filtering a signal, determining that computing the optimal coefficients exceeds the computational capabilities of the first processor, notifying a second processor that computing the optimal coefficients exceeds the computational capabilities of the first processor, and computing, by the second processor, the optimal coefficients. The system and method account for limited computational resources allocated to certain processors in a telecommunications system.

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: 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 acoustic echo canceller comprising an adaptive filter, a post filter, and an adjustable filter. The adaptive filter receives a reference signal and a near-end signal, cancels a portion of the echo in the near-end signal using the reference signal, and provides an intermediate signal having remaining echo not canceled by the adaptive filter. The post filter calculates the cross-correlation between the near-end signal and intermediate signal, and provides a set of coefficients based on a variable suppression parameter. The adjustable filter processes the intermediate signal based on the coefficients and provides an output signal having at least a portion of the remaining echo removed. The acoustic echo canceller may further comprise a channel control unit that provides control signals, an adjustable amplifier, a noise estimator that estimates the noise in the intermediate signal, and a noise reinsertion unit that reinserts a version of the estimated noise back into the output signal.

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: Echo canceling method including the steps of determining an Euclidean norm of an echo-replica signal, determining an Euclidean norm of an input signal, determining a gradient step size, correcting the coefficients of an adaptive filter, deriving an updated echo-replica signal, and determining an updated error signal by subtracting the updated echo-replica signal from the input signal.

Abstract: An echo processor includes an echo suppression amount decider which compares a state of an outgoing input signal to be sent including an echo with a state of an incoming input signal and calculates an echo suppression amount based on the comparison, a time-frequency transformer which transforms the outgoing input signal in the time domain into the frequency domain, thereby generating an amplitude spectrum and a phase spectrum, a noise-spectrum estimator which estimates a noise spectrum from the amplitude spectrum, a mixer which mixes an amount of the noise spectrum based on the echo suppression amount with the amplitude spectrum, and a frequency-time transformer which generates an outgoing output signal including a noise in the time domain on the basis of the amplitude spectrum, with which an amount of the noise spectrum is mixed by the mixer, and the phase spectrum, and outputs the outgoing output signal.

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: A hybrid circuit within a full-duplex transceiver transmits an outgoing signal outward on a communication channel at the same time it receives an incoming signal arriving via the communication channel, and the outgoing and incoming signals sum to form a combined signal. The hybrid circuit generates both the outgoing signal and a replica of the outgoing signal in response to an input signal, and then subtracts the replica from the combined signal in producing a received signal. The received signal includes a component derived from the incoming signal and a residual echo component having peaks resulting from a phase difference between the outgoing signal and its replica. The transceiver periodically digitizes the received signal to produce a data sequence representing the incoming signal.

Abstract: A robust adaptive filter for use in a network echo canceller or other digital signal processing application utilizes a coefficient vector update device that, through the application of fast converging algorithms to a fast impulse response filter yields fast convergence of the adaptive filter's characteristics with the avoidance of divergence due to the onset of double talk. Robustness is also provided, via an adaptive scale non-linearity device which applies an adaptive scale non-linearity to the filter algorithms fed to the fast impulse response filter by the coefficient vector update device, so that the samples of an echo signal to be cancelled which are taken during the onset of double talk can be handled in such a manner that after the double talk detector causes adaptation to cease, the initial, potentially disturbing samples do not cause significant divergence in the filter system.

Abstract: A method and system for performing diagnostic tests on a communications system. Diagnostic test signals are generated by a transceiver included within the communications system. Diagnostic data signals generated within the communications system in response to the diagnostic test signals are collected by the transceiver. The communications transceiver may operate alone, with an assisting communications transceiver, or with a non-assisting communications transceiver. The collected diagnostic data signals are stored and made available for later analysis. The method and system are used to generate and store ECHO Crosstalk, Near End Cross Talk (NEXT) and Far End CrossTalk (FEXT) data signals.

Abstract: An echo canceler and method for a communications system include an echo-containing signal input for receiving a signal and an echo-causing signal source for developing an echo-causing signal. A first filter coupled to the echo-containing signal input and the echo-causing signal source filters the echo-containing signal over a predetermined time period using an existing filter coefficient set to provide a first echo-canceled output signal at a first filter output mode. A trial filter is coupled to the echo-containing signal input and the echo-causing signal source for processing the echo-containing signal over a predetermined time period using a trial filter coefficient set to provide a trial echo-cancel output signal at the trial filter output mode.

Abstract: A non-linear processor for managing a speech signal is disclosed. The non-linear processor includes an adaptive lattice filter that generates a prediction error from a residual echo of the speech signal, a signum non-linearity unit that generates an excitation component from the prediction error, an excitation signal mixer that generates an excitation signal from the excitation component and a noise component, an inverse lattice filter that generates a modified residual echo from the excitation signal, and a gain adjuster that adjusts a power of the modified residual echo.

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: Methods and systems pertaining to preventing the corruption of filter coefficients due to near end speech in an echo canceller that uses an adaptive filter to create a local replica of an echo signal are provided. In accordance with a preferred method, a far end signal is received and the presence of near end speech is indicated. A filter coefficient vector is updated, and copies of the filter coefficient vector are periodically stored. A stored copy of the filter coefficient vector is recovered for use in the echo canceller. Further, an improved echo canceller that prevents the corruption of filter coefficients due to near end speech, having a near end speech indicator, a first buffer, a filter, and a second buffer connected to the filter, is provided. The first buffer receives a far end signal. The filter is responsive to the first buffer and the near end speech indicator.

Abstract: Methods and apparatus compensating for background noise associated with a transmitted signal. In a first embodiment, an adder combines an echo estimate with the the transmitted signal to form a processed signal, a noise reduction circuit uses a background noise estimate to subtract the background noise from the processed signal, and a non-linear processor uses the background noise estimate to combine a compensating background noise signal with the proccssed signal. In another embodiment, the noise reduction circuit divides the processed signal into sub-bands. With variations of an embodiment, the nonlinear processor is a sub-band non-linear processor, the processed signal is divided into sub-bands, and the background estimate includes sub-band noise estimates.