Abstract: The present invention discloses a digital-to-analog conversion apparatus having current source measuring mechanism. A digital-to-analog conversion circuit in turn sets one of thermo-controlled current sources as an initial current source to operate according to two specific input codewords included in an input digital signal to generate an output analog signal. The values of the output analog signal corresponding to the two specific input codewords have opposite signs and the same absolute value. An echo transmission circuit processes the output analog signal to generate an echo signal. An echo-canceling circuit processes the input digital signal according to echo-canceling coefficients to generate an echo-canceling signal and receives an error signal to converge the echo-canceling coefficients.

Abstract: A communication device and an output sidetone adjustment method thereof are disclosed. The method includes the steps of: receiving a far-end sound; receiving a near-end sound to form a sidetone and obtaining an average input volume of the near-end sound; detecting ambient noise of the near-end sound; analyzing an average noise volume and a noise frequency band of the ambient noise to determine whether the average noise volume of the ambient noise is higher than the average input volume; if yes, adjusting the volume of the sidetone in a specific frequency band by increasing it by a first gain value to form an adjusted sidetone, wherein the specific frequency band and the noise frequency band of the ambient noise belong to the same frequency band or different frequency bands; and outputting the adjusted sidetone and the far-end sound.

Abstract: A sound signal processing device, a sound signal processing method, and a sound signal processing program can remove echo signals included in input signals from a plurality of microphones with a simple circuit configuration are provided. The present device 1 includes an output part 20 that outputs a reception signal s2, an input part 30 that combines signals input from each of a plurality of microphones 3 and generates a transmission signal s4, wherein the microphones 3 collect an echo component of the reception signal and voice of a speaker and generate an echo signal es corresponding to the echo component and a sound signal s1 corresponding to the voice of the speaker, a removal signal generator 70 that generates a removal signal s5 removing the echo signal included in the transmission signal based on a filter coefficient F, a controller 50 that calculates the filter coefficient, and a removal part 80 that generates an echo removal signal s6 based on the transmission signal and the removal signal.

Abstract: A communication device and an echo cancellation method are provided. A digital echo canceller is coupled to a transmitting circuit and a receiving circuit to generate an echo energy indicator according to a digital output signal and a digital input signal. A transceiving front-end circuit receives the analog output signal and generates a hybrid interface signal. A hybrid fine-tune circuit generates a first and a second capacitance calibration signals according to the echo energy indicator. An analog echo cancellation circuit receives the first and second capacitance calibration signals, and includes a first and a second variable capacitors controlled by the first capacitance calibration signal and a third and a fourth variable capacitors controlled by the second capacitance calibration signal. The analog echo cancellation circuit receives the analog output signal and the hybrid interface signal, and generates the analog input signal according to the first and second capacitance calibration signals.

Abstract: An example apparatus provides an input signal based on sound waves received by one or more microphones. The input signal includes a voice command component and one or more interference components. The apparatus receives audio data over one or more computer networks and the audio data corresponds to the one or more interference components. The apparatus uses the audio data to remove a portion of the one or more interference components from the input signal to generate an output signal, and provides the output signal, as an estimate of the voice command component, for speech recognition.

Abstract: A method for avoiding an offset of a membrane (3) of an electrodynamic acoustic transducer (1) having two voice coils (7, 8) is presented, wherein a control voltage (UCTRL) is applied to at least one of the voice coils (7, 8) until the electromotive force (Uemf1) of the first coil (7) or a parameter derived thereof and the electromotive force (Uemf2) of the second coil (8) or a parameter derived thereof substantially reach a predetermined relation. Furthermore, an electronic offset compensation circuit (12) is presented, which performs the above application of a control voltage (UCTRL). Finally, the invention relates to a transducer system with a transducer (1) and an electronic offset compensation circuit (12) connected to the transducer (1).

Abstract: A voice processing device includes a first sound collecting unit for generating a first voice signal; a human-body vibration obtaining unit for generating a human-body vibration signal; a first distance calculating unit for calculating a ratio in power or amplitude between the human-body vibration signal and the first voice signal and for calculating a first distance from the first sound collecting unit to a sound source in accordance with the ratio and distance estimation information; a second distance calculating unit for calculating, for each of a plurality of frequencies, a second distance from the first sound collecting unit to a sound source which produces a component of a frequency of a first frequency signal; a gain determining unit for determining, for each of the plurality of frequencies, a gain based on a comparison result between the first distance and the second distance.

Abstract: Systems and methods for controlling adaptivity of noise cancellation are presented. One or more audio signals are received by one or more corresponding microphones. The one or more signals may be decomposed into frequency sub-bands. Noise cancellation consistent with identified adaptation constraints is performed on the one or more audio signals. The one or more audio signals may then be reconstructed from the frequency sub-bands and outputted via an output device.

Abstract: Generally, this disclosure provides systems, devices, methods and computer readable media for an audio device with noise interference mitigation capability. The audio device may include an interference detection module to detect an interfering voice audio signal received from an interfering audio source. The device may also include an interference identification module to identify the interfering audio source. The device may further include an impact estimation module to estimate an impact of the interfering voice audio signal on a user of the audio device and a communication module to transmit a volume correction request to the interfering audio source. The volume correction request may be based on the identification and the estimated impact. Other embodiments may be described and claimed.

Abstract: A power amplifier circuit including a power transistor, a variable impedance circuit, first and second envelope detecting circuits is provided. The power transistor receives an input signal and outputs an output signal. The variable impedance circuit includes an impedance control transistor and a first filter capacitor. An equivalent impedance of the variable impedance circuit varies with the input signal. The impedance control transistor is coupled to the power transistor and receives a control voltage. The first filter capacitor is coupled between the impedance control transistor and ground. Both the first and the second envelope detecting circuits detect the input signal to dynamically control the equivalent impedance of the variable impedance circuit.

Abstract: The present disclosure illustrates an acoustic echo cancellation method and system using the same. The acoustic echo cancellation method comprises the following steps. Firstly, a prior-knowledge matrix comprising a plurality of space vectors is built. Then, an initial filter vector is generated by the prior-knowledge matrix and an initial weighting vector. The weighting vector is updated based on the difference of the echo signal and the estimated signal in an iteration algorithm, and the coefficient of the filter vector is updated according to the updated weighting vector. An estimated signal is generated according to the updated filter vector and the original signal. Finally, the next echo signal is cancelled by the near-end estimated signal.

Abstract: Preventing unintended distribution of audio information may comprise analyzing audio data of a speaker's speech received by a microphone; determining automatically by a processor, from the analyzing whether the speaker's speech is intended to be distributed to an audience via the microphone; and in response to determining that the speaker's speech is not intended to be distributed to the audience via the microphone, performing one or more actions.

Abstract: Preventing unintended distribution of audio information may comprise analyzing audio data of a speaker's speech received by a microphone; determining automatically by a processor, from the analyzing whether the speaker's speech is intended to be distributed to an audience via the microphone; and in response to determining that the speaker's speech is not intended to be distributed to the audience via the microphone, performing one or more actions.

Abstract: In at least some embodiments, a computer system includes a processor and a network interface coupled to the processor. The computer system also includes a system memory coupled to the processor, the system memory storing a communication application having a conferencing user interface. The conferencing user interface, when executed, enables a user to set up a conferencing session by selecting one of a plurality of conferencing environment templates (CETs). Each CET is matched to one of a plurality of different sets of acoustic echo cancellation (AEC) parameters to be applied during the conferencing session.

Abstract: A power line carrier communication apparatus which prevents leakage of an electromagnetic wave and stably operates while employing a simplified configuration is provided. The power line carrier communication apparatus may include a transmission circuit which is connected to a power line and which transmits a signal to the power line; a reception circuit which is connected to the power line and which receives the signal from the power line; and an output suppression circuit which is connected to the power line and which, by connecting a signal attenuating element between the transmission circuit and a common only when the transmission circuit transmits the signal, suppresses an output of the signal transmitted from the transmission circuit.

Abstract: Methods, systems, and apparatus for controlling audio signal processing are described herein. In accordance with certain embodiments, a measure of coupling between at least one microphone and one speaker associated with an audio device is obtained. The measure of coupling may be used to control audio signal processing, such as sidetone generation and application of multi-channel decorrelation to the audio signal.

Abstract: Systems and methods for adaptive power control are provided. A minimum power level of a primary acoustic signal is estimated. The minimum power level may then be compared to at least one power threshold. Subsequently, a large power consuming system is controlled based on the comparison of the minimum power level to the power threshold.

Abstract: A vibrator designed to be integrated into a portable electronic device as part of a haptic alert device has a linear electric motor that is coupled to actively drive a moveable weight in back-and-forth directions to produce the desired vibration. A lock mechanism has a restraining member that is biased into a lock position, which prevents the moveable weight from moving in the back-and-forth directions when the vibrator is inactive. Other embodiments are also described and claimed.

Abstract: The present invention relates to a method as well as to a computing device (20) for editing a noise-database (13) containing noise information, said noise information being derived from noise signals within an audio stream (19). In order to enhance possibilities to create and utilize context information which emerge from tracking noise signals from an audio stream, for example a telephone call, the above method is characterized by the following steps: A) in a localizing step (14), determining geographical data of the location the noise signals origin from; B) in an analyzing step (15), analyzing the noise signals with reference to the noise content; C) in a linking step, linking the analyzed noise signals to said geographical data to create noise information; D) in a storing step, storing said noise information within said noise-database (13).

Abstract: An echo signal processing apparatus is disclosed. The echo signal processing apparatus is utilized for generating a cancellation signal by using group phenomenon of a frequency response of an echo signal to remove the echo signal. The echo signal processing apparatus has lower cost and is able to remove the echo efficiently.

Abstract: Microphone arrays (MAs) are described that position and vent microphones so that performance of a noise suppression system coupled to the microphone array is enhanced. The MA includes at least two physical microphones to receive acoustic signals. The physical microphones make use of a common rear vent (actual or virtual) that samples a common pressure source. The MA includes a physical directional microphone configuration and a virtual directional microphone configuration. By making the input to the rear vents of the microphones (actual or virtual) as similar as possible, the real-world filter to be modeled becomes much simpler to model using an adaptive filter.

Abstract: The sidetone noise level in a wireless or wired telephone, for example, is reduced by adjusting both noise and signal levels and/or by improving the signal-to-noise ratio of such signals.

Abstract: An adaptive audio system can be implemented in a communication device. The adaptive audio system can enhance voice in an audio signal received by the communication device to increase intelligibility of the voice. The audio system can adapt the audio enhancement based at least in part on levels of environmental content, such as noise, that are received by the communication device. For higher levels of environmental content, for example, the audio system might apply the audio enhancement more aggressively. Additionally, the adaptive audio system can detect substantially periodic content in the environmental content. The adaptive audio system can further adapt the audio enhancement responsive to the environmental content.

Abstract: An HVAC unit includes a transformer and a system controller. The transformer is configured to receive power from a first and a second phase of a three-phase power source and to produce a first reduced-voltage waveform therefrom. The system controller is adapted to sample the reduced voltage waveform to determine a figure of merit of the three-phase power source. The controller is further configured to operate the HVAC unit in response to the figure of merit.

Abstract: The present invention relates to an electronic device that can output received voice and input transmitted voice at the same time and enhances a suppression function for echo due to the received voice sneaking into the transmitted voice. An electronic device (e.g., portable telephone terminal device) outputs the received voice from a voice output unit (speaker), inputs the transmitted voice through a voice input unit (microphone), and includes an echo canceller unit that subtracts a pseudo echo signal for the received voice from the transmitted voice to suppress an echo component in the transmitted voice and a controlling unit that changes an echo suppression amount of the echo canceller unit in accordance with the received voice volume.

Abstract: The present invention is related to methods and systems for reducing or canceling echoes during fax communications. In an embodiment, during a fax call, a first fax signal from a first fax device intended for a second fax device is detected. Based at least in part on detecting the first fax signal from the first fax device or on the termination of the first fax signal, an echo audio packet is muted, wherein the echo audio packet is at least partly the result of the first fax signal being transmitted over a network, the network including at least a data network.

Abstract: Constant phone-call quality is kept irrespective of setting of a tone adjusting function of a cellular phone under connection. There is provided a sound collector for collecting voices and converting the voices to a voice signal, a communication unit for communicating with a cellular phone terminal according to a predetermined communication protocol, and a reproducing unit for reproducing the voice signal received in the communication unit.

Abstract: An enhanced sidetone system is disclosed which provides the user of a telecommunications terminal, while speaking, with immediate audio feedback that corresponds to what the far-end party is probably hearing. The sidetone system continuously samples the input speech signal from the user and also obtains signal quality statistics of the transmission path. These statistics can include descriptions of network quality-of-service characteristics (e.g., packet loss rate, etc.) and/or media quality characteristics (e.g., audio distortion due to echo cancellation, etc.). These statistics enable the disclosed technique to determine whether the transmitted signal quality is acceptable. When an unacceptable condition in transmit-path signal quality is detected, the technique modifies the traditional (main) sidetone signal.

Abstract: In a communication machine room wideband noise suppression system, a sensing unit detects a noise source produced by a fan during the operation thereof and generates a noise input signal and a feedback signal, which are sent to a signal amplifying unit for signal amplification. The amplified signals are then sent to a signal converting unit and converted into digital signals. A multi-channel hybrid controller receives the digital signals and makes corrections and conduct rapid convergence algorithm to derive a reverse digital signal, which is sent to the signal converting unit and converted into a reverse analog signal. The reverse analog signal is sent to the signal amplifying unit for power amplification to generate a control signal for driving a loudspeaker unit to produce interfering acoustic wave, so as to cancel out the noise source and achieve the purpose of eliminating wideband noise.

Abstract: A method, device, and computer-readable storage medium for providing digital side tone are disclosed. These embodiments are directed toward processing digital side tone in a secondary processing unit external to a primary processing unit that includes the main digital signal processor, thereby reducing delay in producing side tone and freeing up processing resources in the primary processing unit. In an environment where the sampling rate of a digital-to-analog converter in the secondary processing unit is an integer multiple of the sampling rate of an analog-to-digital converter in the secondary processing unit, the components of the secondary processing unit are made to operate such that the digital side tone signal and the main tone signal are equal in sampling frequency and number of samples.

Abstract: An audio system including an audio-enhancing station including at least one speaker and an audio-enhancing station docking connector, a music phone configured to play audio files and function as a wireless communication device separate from the audio-enhancing station, the music phone including at least one speaker to emit sound from the playing of the audio files, and a music phone docking connector that connects with the audio-enhancing station docking connector to dock the music phone with the audio-enhancing station, wherein the music phone and the audio-enhancing station together form an audio system where the at least one speaker of the music phone and the at least one speaker of the audio-enhancing station emit sound from the playing of the audio files on the music phone.

Abstract: A system and method are provided for detecting acoustic echo in a telecommunications transmission. Detected acoustic echo is analyzed to determine if the acoustic echo includes characteristics indicative of use of a speakerphone by a remote party. Specific characteristics analyzed include the presence of multiple bulk delay values, echo magnitude, and the length of time for echo delays present within the telecommunications system. If the detected acoustic echo falls outside the range of acceptable parameters, or otherwise exceeds threshold limits, a warning signal is transmitted to the user indicating a high likelihood that the remote party is using a speakerphone. Warning the user of speakerphone use by the remote party can be helpful in ensuring that the communications between the parties is maintained in confidence.

Abstract: A side-tone circuit comprises a microphone configured to generate a reverse link audio signal and a side-tone noise suppressor. The side tone noise suppressor is configured to receive the reverse link audio signal, suppress noise in the received reverse link audio signal, and output a noise suppressed side-tone signal. The side-tone noise circuit also includes a speaker configured to receive the noise suppressed audio signal and communicate it to a user.

Abstract: An enhanced sidetone system is disclosed which provides the user of a telecommunications terminal, while speaking, with immediate audio feedback that corresponds to what the far-end party is probably hearing. The sidetone system continuously samples the input speech signal from the user and also obtains signal quality statistics of the transmission path. These statistics can include descriptions of network quality-of-service characteristics (e.g., packet loss rate, etc.) and/or media quality characteristics (e.g., audio distortion due to echo cancellation, etc.). These statistics enable the disclosed technique to determine whether the transmitted signal quality is acceptable. When an unacceptable condition in transmit-path signal quality is detected, the technique modifies the traditional (main) sidetone signal.

Abstract: A second output transmission signal (“TX2”) added to a line driver is a scaled version of the main output transmission signal (“TX1”). TX2 is scaled from TX1 by a variable scale factor K. An adaptive hybrid circuit subtracts TX1 and TX2 from a line signal carrying both a line transmission signal and a line received signal (“RX”). A programmable impedance Ztune is coupled between the TX2 output of the line driver and the RX output of the adaptive hybrid circuit. A transmission echo in the output RX signal is measured. K and Ztune are then adaptively tuned to minimize the transmission echo. The hybrid in this case becomes a 4-port network, one port specifically added to adaptively cancel the transmission echo in the RX output of the adaptive hybrid circuit. Alternatively, the hybrid may be a 3-port hybrid including variable impedances to cancel the line transmission signal.

Abstract: A real loop and method for using the loop for testing sidetone signal level are provided. The loop is modeled from a coil of cable to represent a straight wire loop. Several conducting wires in a cable may be used. The cable may be configured in a coil and may have a first end and a second end exposed for electrical connections to an even number of wires. Each wire in the even number of wires may be capable of being identified as the same wire on each of the first and second ends of the cable. One or more pairs of wires in the even number of wires may be electrically connected to form an electrical loop. An electrical current passes in opposite directions in the wires in each pair of wires.

Abstract: Disclosed are a method and an apparatus for processing communication signals in a telephony or communication system. In various embodiments, the method or the apparatus comprises identifying a first signal which comprises an incoming communication signal and a feedback signal, controlling a logic for cancelling or suppressing a feedback effect based at least in part upon the first signal, processing the first signal using an adaptive filter based at least in part upon the first signal and a result of controlling the logic for cancelling or suppressing the feedback effect, and causing the processed first signal to be transmitted to a first output device. In some embodiments, the method and the apparatus further comprise identifying a communication context which the communication system is currently in. In some embodiments, the method and apparatus further comprise determining a filter coefficient for the adaptive filter.

Abstract: An ADSL POTS splitter includes low pass filter. The low pass filter includes a first inductor portion in communication with the line interface and a second inductor portion isolated from the line interface via an isolation capacitor. In one embodiment, the first inductor portion is characterized for non-saturation in conventional operating ranges. The first non-isolated inductor portion has a respective DC saturation current and the isolated inductor portion has a respective DC saturation current. The DC saturation current of the first non-isolated inductor portion is substantially greater than the DC saturation current of the isolated inductor portion for achieving higher inductance in the isolated inductor portion.

Abstract: The present invention relates to a transceiver for bidirectional frequency division multiplexed transmission, a communication system including one or more transceivers. Optionally, the communication system is a communication system for a digital subscriber line. The transceiver comprises transmission means with a voltage source output or a current source output for transmitting data in a transmission frequency range, receiving means for receiving data in a receiving frequency range, and a coupling impedance for connecting the transmission means and the receiving means to a transmission medium. The magnitude of the coupling impedance in the transmission frequency range is smaller than the magnitude of the coupling impedance in the receiving frequency range if the transmission means has a voltage source output and is higher than the magnitude of the coupling impedance in the receiving frequency range if the transmission means has a current source output.

Abstract: There is disclosed an echo canceller circuit for use in a full duplex transceiver of the type comprising a line driver capable of sending analog transmit signals through a cable and comprising a line receiver capable of receiving analog receive signals from the cable. An echo canceller impedance model circuit is coupled to an output of the line driver and is coupled to an input of the line receiver. The echo canceller impedance model circuit generates an echo canceller current that is equal in magnitude and opposite in phase to a current that represents signal echoes that are present in the analog receive signals. The echo canceller impedance model circuit has a variable impedance for generating the echo canceller current. The variable impedance has at least one variable resistor and at least one variable capacitor.

Abstract: A wireless communication device is disclosed that provides a sidetone to the device user. The device converts an outbound analog audio signal to an outbound audio bitstream from which a sidetone bitstream is extracted. The device also converts an inbound digital audio signal to an inbound audio bitstream. A filter in the device both adds the sidetone bitstream to the inbound audio bitstream and filters the resultant added bitstreams to provide an analog audio signal with sidetone.

Abstract: A digital telephone answering device having speakerphone capability allows a recorded message to be played back and heard by the far-end party as well as over the local speakerphone by the near-end party during speakerphone operation as if it were a normal receive signal. Moreover, normal speakerphone conversation is possible during this conversational playback mode allowing the far-end party and/or near-end party to break-in over the played back pre-recorded message as desired and be heard by the other party. Both message and conversation signals are preferably (but not necessarily) at similar levels. Also, the message playback at the near end is preferably (but not necessarily) subject to the same speakerphone digital volume control as that received at the far end.

Abstract: A combined sidetone and hybrid balance apparatus and method of operating same are disclosed. An embodiment of the present invention may provide both a hybrid balance mode of operation and a sidetone generation mode of operation within a single integrated circuit device. The functionality provided by an embodiment of the present invention may be used in both Internet protocol (IP)-based telephones and residential gateways, and may be incorporated within the IP telephone chip used in such devices. An embodiment of the present invention may be used to reduce processor demand and the cost of materials, permitting greater IP telephone or residential gateway functionality at lower cost.

Abstract: A hybrid circuit is disclosed for effectively communicating information over a telecommunications line to which a transmitter and receiver are connected. The hybrid circuit is configured as a filter for filtering at the receiver input signals at predetermined frequencies appearing on the transmitter output and the telecommunications line. The filter circuit additionally forms a capacitive divider for scaling signals appearing at the transmitter output and canceling the scaled signals at the receiver input with related signals appearing on the telecommunications line. In one embodiment of the present invention for communicating using the ADSL protocol, the filter circuit is a first order high pass filter.

Abstract: A system and method for adjusting the volume level of a communications device in response to a voice volume/ambient noise relationship. Ambient noise and voice volume are sampled and compared to a predetermined ambient noise to voice volume relationship. Voice volume is adjusted up or down in response to a control signal that is generated in view of the comparison between the sampled voice volume to ambient noise relationship and the predetermined ambient noise to voice volume relationship.

Abstract: A line interface couples a data transceiver to a transmission line via a transformer, the data transceiver transmitting signals in a first frequency range and receiving signals in a second frequency range. The line interface includes an input port for receiving an input signal voltage from an analog front end (AFE) chip, an output port, a line driver for amplifying the input signal voltage and supplying a transmit signal to the output port, a line port for sending the transmit signal and receiving a receive signal, termination resistors coupled between the output port and the line port, a receive signal port for supplying the receive signal to the AFE chip, a receive amplifier formed on the AFE chip coupled to the receive signal port, and a bridge network resistively coupling the line port and the output port to the receive signal port, the bridge network having a low-pass filter characteristic.

Abstract: A full duplex telephone set for hearing impaired users includes a digital signal processor incorporating a pair of adaptive transversal filters to cancel line and acoustic echo feedback from the tip and ring signals of the set, and to condition and control the amount of a replacement side tone fed back to the receiver. This enables at least 50 dB of gain to be applied to the modified received signal without initiating unstable operation, or howl, in the set, and ensures that the received and side tone signals are heard by the user with equal comfort and clarity. The digital signal processor also enables additional enhancements to be made to the modified received signal that are beneficial to hearing impaired users, such as dynamic range compression, “soft” limiting, and low level emphasis within selectable frequency bands.

Abstract: Method and apparatus for detecting far end speech in a telephone apparatus. A speaker (14) produces far end sound from a far end signal, when received by the telephone apparatus. A microphone (12) generates a near end signal from near end sound, for transmission from the telephone apparatus. A hybrid (24) which passes the near and far end signals from an external two wire link (30) to and from the speaker and microphone, on respective first and second paths (1, 2) of a four wire link (28). Muting of the near end signal is effected when signal derived from the first path (1) of the four wire link (28) is detected as being representative of a condition of existence of far end signal.

Abstract: The variable sidetone system provides an enhanced method for inducing the subscriber maintain the amplitude of their voice at an appropriate level by exaggerating the amplitude characteristics of the sidetone returned to the subscriber. Rather than have the sidetone level linked to the amplitude of the subscriber's voice signal in a linear non-adjustable 1:1 relationship (as is done in the prior art), the present variable sidetone system implements an adjustable relationship between the amplitude of the subscriber's voice signal and sidetone levels, thereby allowing the variable sidetone system to provide the necessary acoustic feedback to subscribers to maintain the subscriber's voice signal within the predetermined bounds of the voice encoder.

Abstract: There is disclosed an echo canceller circuit for use in a full duplex transceiver of the type comprising a line driver capable of sending analog transmit signals through a cable and comprising a line receiver capable of receiving analog receive signals from the cable. An echo canceller impedance model circuit is coupled to an output of the line driver and is coupled to an input of the line receiver. The echo canceller impedance model circuit generates an echo canceller current that is equal in magnitude and opposite in phase to a current that represents signal echoes that are present in the analog receive signals. The echo canceller impedance model circuit has a variable impedance for generating the echo canceller current. The variable impedance has at least one variable resistor and at least one variable capacitor.