Abstract: A line interface couples signals between a data transceiver and a transmission line having a load impedance Z. The line interface includes a transformer, a driver circuit for supplying a transmit signal from the data transceiver to the transformer, and a receiver circuit for receiving a receive signal from the transformer. The transformer includes a first port coupled to the transmission line, a second port coupled to the driver circuit, a third port coupled to the receiver circuit, a first winding part having a turns ratio of 1: n, where n>1, for coupling the transmit signal from the second port to the first port, and a second winding part having a turns ratio of 1: m, where m<n, for coupling the receive signal from the first port to the third port.

Abstract: A telephone receiver circuit with sidetone signal generation controlled by voice activity detection in accordance with the present invention uses the voice activity detector (VAD) to detect the presence of voice activity within the microphone signal and dynamically adjust the sidetone signal generation to compensate for noisy environments by eliminating or reducing the sidetone signal in the absence of voice activity. Hence, a sidetone signal is generated in the presence of voice activity, when feedback is required, while the sidetone signal is not generated in the absence of voice activity, since audio feedback for the user is not required then.

Abstract: A filtering circuit is described for filtering out meter pulses of analog telephone communication, for example in analog modems. The circuit is mainly composed of resistors and capacitors and is configured as a double T element or a Wien bridge circuit. The filtering circuit is needed for attenuating interferences to such an extent that integrated circuits operating with low voltages are not overloaded.

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

Abstract: The present invention provides a passive electronic filter circuit for telephony equipment used in conjunction with xDSL that reduces the number of discrete components over existing filters. In one exemplary embodiment the present invention employs a POTS filter having a coupled-inductor array made up of two cascaded pairs of coupled inductors that share one common ferrite core. The use of a common ferrite core reduces the number of coupled inductor packages required to perform the same filtering.

Abstract: An apparatus and method for interfacing a communication device, such as a Customer Premise Equipment (CPE) to different telephone networks. The telephone networks typically provide a voltage of 48 volts to the CPE. However, this DC voltage tends to fluctuate. Furthermore,the impedances of the telephone network varies. The apparatus provides a constant current to the communication device while coupled to different telephone networks. Since most communication devices use amplitude modulations for transmitting data, a constant DC biasing current is desired. The apparatus can sense load impedances of the telephone networks, and dynamically adjust the current source so that DC biasing currents remain relatively constant. The apparatus is coupled to the telephone network so that communication between the communication device and the telephone network is optimized.

Abstract: Techniques for software configurable sidetone are provided. A user can select one or more characteristics of the sidetone and the sidetone signal will be generated according to an input signal and the user selected characteristic or characteristics. A bipartite calculation can be utilized to generate the sidetone such that the sidetone increases by a decreased percentage of the input signal above a threshold. Additionally, characteristics of multiple configurations can be stored in a configuration database for use with each configuration.

Abstract: A system for matching impedance in the circuitry of an xDSL communication device to improve noise cancellation resulting from signal leakage between transmit and receive signals. The system for matching impedance includes a transformer configured to couple an impedance, which is substantially equal to a line impedance, to a line coupling transformer, and is applicable to all known hybrid topologies. The transformer is ideally as closely matched to the line coupling transformer as possible. This technique allows greatly improved impedance matching in the hybrid, which directly benefits the performance of xDSL communication devices.

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: A side-tone control unit for a telecommunication instrument is configured to include a side-tone amplifier having a transfer function selectively controlled in accordance with a set of amplifier parameters, and a side-tone controller, coupled to the side-tone amplifier, for selectively applying the set of amplifier parameters to the side-tone amplifier based on the detected energy of an uplink signal and the detected energy of a downlink signal. The side tone controller can select a particular set of amplifier parameters by indexing a table in accordance with quantized values of the detected uplink signal energy and the detected downlink signal energy.

Abstract: A current sense circuit for sensing a current in a subscriber loop comprising a transformer having a first and second winding coupled for sensing a differential current in a loop, a sense winding for producing a sense current in response to a magnetic flux induced in the transformer due to the differential current, an amplifier circuit for receiving the sense current and wherein the sense current is indicative of a signal on the loop, and the amplifier being configured such that a transformer introduces a low insertion loss at a frequency of operation of the loop.

Abstract: An audio processing module, in accordance with the invention, includes an input for receiving input signals. A side tone generator is included for receiving the input signals and for generating a side tone to be output. A noise suppressor is coupled to the input for suppressing noise of the input signal prior to a coding process, the noise suppressor providing feedback to the side tone generator to adjust a gain of the input signal to the output. A method for adjusting the gain of the side tone generator is also included.

Abstract: A system comprising a base unit and a remote unit for transmitting telephonic data on AC power lines is disclosed. The base unit includes a transmitter section, a receiver section, a power line interface, a telephone line interface, a power supply, and a first microcontroller. The telephone line interface is coupled to a telephone line to apply the telephonic data signals to and receive signals from the telephone line. The telephone line interface has a ring detector, a muting circuit, and a hook switch. The transmitter is coupled to the telephone line interface and the power line interface. The transmitter has a modulator, an oscillator, an amplifier, and a filter. The receiver is coupled the telephone interface and the power line interface. The receiver has a filter, an amplifier, and a detector. The remote unit includes a transmitter section, a receiver section, a power line interface, a telephone interface, a power supply, and a second microcontroller.

Abstract: A known cross-talk adaptive noise cancellation apparatus uses two adaptive filters for suppressing the correlated noise in the main signal, which contains wanted voice and unwanted noise, by a reference signal, which contains unwanted noise and wanted voice via cross-talking. A noise canceler is provided in the present invention including some additional parts, such as a voice detection circuitry and a step size decision circuitry. The voice detection circuitry detects whether the wanted voice exists. The step size decision circuitry decides what the step sizes are used for the next update of the two adaptive filters.

Abstract: A method and apparatus for a telephone line interface or data access arrangement (DAA) which includes a shunt regulator in series with a line modulator. A first sense resistor is placed in series between the shunt regulator and the line modulator to provide a measurement of the amount of current distortion in the DAA shunt regulator. The voltage across the first sense resistor is fed back to the line modulator. A second sense resistor is inserted into the line modulator circuitry to provide a measurement of the system current distortion that is outside of the path containing the sense resistor used to monitor the current distortion in the DAA shunt regulator. The voltage across the second sense resistor is also fed back to the line modulator. The line modulator is capable of adjusting the AC modulation and the DC termination presented to the telephone line.

Abstract: A method and system for providing suppression of near-end speech energy for tone signal detectors. In accordance with the invention a system or device is connected between the tip and ring on a subscriber loop, the subscriber loop being connected to office equipment and that subscriber's station set. The system also includes a receive interface that is connected to either a tone signal detector or a frequency shift keying modulator. The system achieves speech cancellation by forming a Wheatstone bridge with a mirror circuit possessing a scaled image of the loop and office equipment impedance. In one embodiment of the invention the Wheatstone bridge is formed using the combined impedance of the loop and office equipment as the impedance that is to be matched by a set of balanced networks included within the system.

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

Abstract: An interface for coupling a modem port with a transmission line includes a hybrid converter having a variable gain amplifier coupling a transmit path of a first differential polarity and a received path of a second differential polarity.

Abstract: A side-tone control unit for a telecommunication instrument is configured to include a side-tone amplifier having a transfer function selectively controlled in accordance with a set of amplifier parameters, and a side-tone controller, coupled to the side-tone amplifier, for selectively applying the set of amplifier parameters to the side-tone amplifier based on the detected energy of an uplink signal and the detected energy of a downlink signal. The side tone controller can select a particular set of amplifier parameters by indexing a table in accordance with quantized values of the detected uplink signal energy and the detected downlink signal energy.

Abstract: A method and system for reducing background noise during a telephone call. When a caller begins dialing a telephone number, the system receives an ambient noise level measured near the caller's telephone. The system then computes an inverse noise waveform, which corresponds to the received ambient noise level. The inverse noise waveform is then transmitted along the same telephone line used by the completed call. The inverse noise waveform reduces the ambient background noise from the caller's location for the duration of the telephone call.

Abstract: A low noise, low distortion, power efficient DSL/cable line driver has a double Wheatstone impedance bridge network that functions to prevent contamination of DSL signals that are received by and transmitted from the line driver. The line driver employs an external current sensing impedance that can be either purely resistive or complex and that can be selected to accommodate a particular transmission medium. The line driver further employs an internal programmable resistor that can be programmably adjusted to accommodate changes in transmission medium impedance to optimize sidetone rejection.

Abstract: A telephone with a selectable level of sidetone for use in noisy environments. The selectable level of sidetone is achieved by forming a switch-controlled feedback path including a gain setting resistor from the output of a transmit amplifier in a hybrid circuit to a summing node virtual ground at one input of a differential amplifier driving the handset speaker. The summing node is also coupled to the audio output port of the three port hybrid so as to also receive incoming audio signals from the tip and ring lines. A microcontroller having configuration data controls whether the switch is closed and the feedback path exists or the switch is open and no feedback occurs. Lower sidetone or no sidetone is useful in phones used in noisy environments such as factory floors or restaurants.

Abstract: A 1st telephone apparatus comprises a transmission system for transmitting a transmission voice SIG to a network, a receiving system for receiving a reception voice SIG from the network, a recording system for generating a recording SIG derived from the voice SIGs from the transmission and the receiving system, a memory for storing and reading the recording SIG, and a switch for playing back the reception voice SIG in a communication mode and playing back the recording SIG from the memory in a playback mode. Coding and decoding may performed by a single DSP through a time division manner. A 2nd telephone apparatus comprises a transmission and receiving systems, 1st and 2nd switches, a memory circuit, and a voice presence detector for detecting the presence of the voice SIG of the transmission system, the 1st switch stores the voice SIG from the voice SIG from the transmission system prior to the voice SIG from the receiving system.

Abstract: The invention concerns digital processing of analog signals in a telephone set. A signal path runs between a telephone line and the handset of the telephone. A digital signal processor is interconnected within this signal path, together with appropriate analog-to-digital, and digital-to-analog, converters. This approach allows digital processing of the information carried by the signal path. The digital processing facilitates, for example, (1) a high degree of separation of the handset's microphone signal from the received signal, and (2) independent control of volumes of the received signal and sidetone.

Abstract: A telephone, particularly a portable handset operable in a digital radio telephone system, has a sidetone gain circuit 28 which applies one of two levels of gain G1 G2 dependent on whether a detector 30 detects a high magnitude of input signal 14 corresponding to speech or a low level of input signal 14 corresponding to an idle condition. With varying magnitude of input signal 14, the gain switches between G1 and G2 dependent on two thresholds T1 and T2 of the input signal, the relationship including hysteresis to prevent unduly rapid switching.

Abstract: An electronic voice circuit configuration can be connected through a first and a second line terminal to a telephone line. The electronic voice circuit configuration has a calling part, a transmitting device, a receiver device, a symmetrical modulator circuit with voltage stabilization, a symmetrical equipment impedance device with a device for symmetry control and direct-current limiting, and a symmetrical control device.

Abstract: When telecommunications systems with analog and digital circuit technology are interconnected, the two-wire/four-wire transitions are achieved with hybrid circuits in which mismatches cannot be avoided. They result in signal reflections in the form of interfering echoes which can considerably reduce speech intelligibility. The known methods of echo compensation for hands-free facilities are not suitable for compensating line echoes, since the signal level and the delay times of lines can be very different. A compander (3.7) is used to suppress the effect of line echoes, whose transmission characteristic, in addition to being controlled by a coupling factor between the transmit path and the receive path, and by a control signal determined by the noise level, is also controlled in the invention with the speech level that is received from the distant subscriber.

Abstract: In an apparatus of adjusting a sound level of a reception voice in a telephone, an audio signal including a transmission voice signal and a peripheral noise is inputted through a microphone. A gain control signal corresponding to a level of the peripheral noise is generated while the transmission voice signal is not present and the gain control signal immediately before input of the transmission voice signal is started is held and outputted while the transmission voice signal is present. A variable gain amplifier amplifies a reception voice signal in accordance with the gain control signal such that a voice corresponding to the reception voice signal is outputted from a speaker with a sound level determined in accordance with the gain control signal.

Abstract: The present invention relates to an adaptive telephone interface for use in a telephone apparatus comprising a telephone terminal including signal amplifier means, and audio/input device. The telephone interface is disposable between the terminal and the audio/input device, and comprises a processor for generating a test signal and sending this to the terminal at a plurality of different amplitudes. The processor is further formed and arranged for monitoring signals returned from the terminal by means of sidetone coupling, so as to identify the test signal amplitude at which clipping of at least part of the test signal occurs, and consequentially adjusting the amplification by the interface of a signal received from the audio input device, before sending it on to the terminal so as to avoid substantially clipping thereof by the terminal signal amplifier means in use of the telephone apparatus.

Abstract: A noise elimination scheme is disclosed which enables the prior established telephones or telephone exchange modules linked thereto to eliminate high-frequency noises as applied from outside portable radiotelephone transceiver units. A choke coil (15) is arranged on a circuit board (2) of a telephone main body (1) at a respective one of several nodes for connection with external communication lines (5). Another choke coil (15) is provided at each interconnection node between the circuit board (2) and signal transmission lines (6) of a telephone handset (10). Respective choke coils (15) have a specific stray capacitance of 0.3 pF or below to insure successful elimination of high frequency noises of at or above 800 MHz.

Abstract: A balanced network in a keyphone system has a filtering unit which is composed of: first and second filtering circuits, formed on the transmission path and having an inverted-phase feature, for adjusting the level of the transmission signal and for removing spurious waves thereof; an inverted-phase signal branch point disposed between the first and second filtering circuits, for branching out an inverted-phase transmission signal; an operational amplifier disposed on the reception path, for adjusting a level of the reception signal; a signal superimposing node disposed ahead of an input of the operational amplifier; and a feedback signal level adjusting circuit disposed between the inverted-phase signal branch point and the signal superimposing node, for adjusting the level of the inverted-phase transmission signal which has been branched out from the inverted-phase signal branch point so as to reduce a feedback signal, transmitted from the transmission path to the reception path through the common node, to a

Abstract: The speech circuit matches the impedance of the telephone line by synthesizing a complex impedance using a positive feedback loop which has a single resistor (11), and cancels out the side tone using a subtractor (20') which extracts from the signal (Va) coming from the line a signal (Vb) correlated to the signal to be transmitted. In order to achieve cancellation of the side tone unaffected by the noise produced in the impedance synthesizing circuits, the signal (Vb) is derived by processing the signal present in the resistor (11) at the output of the feedback loop.

Abstract: A speakerphone which employs non-linear amplifiers to compress transmit and receive signals, and level detectors to determine levels of the compressed transmit and receive signals. Compressed transmit and receive signal levels are compared against one another by a comparator with hysteresis, to enable one of two speakerphone modes of operation--either a transmit mode or a receive mode. Fixed attenuators are switched into either the receive signal path or into the transmit signal path, depending upon the current speakerphone mode. The speakerphone functions described herein are readily implemented using inexpensive control circuitry and a standard telephone line interface, which can be shared by a telephone handset and other components of an integrated telephone system.

Abstract: An apparatus and method for operating a speaker telephone in either full-duplex mode or half-duplex mode. In the absence of an earphone connection, the speaker telephone operates in half-duplex mode to prevent undesired feedback of speaker output to microphone input. Upon detecting insertion of an earphone plug into an earphone jack, the speaker telephone transitions to a full-duplex mode of operation. Upon detecting removal of the earphone plug, the speaker telephone reverts to the half-duplex mode of operation. The speaker telephone includes a controller configured to detect an earphone insertion event and enable full-duplex operation in response. The controller enables half-duplex operation upon detecting earphone removal.

Abstract: A speech circuit is disclosed which solves the serious problem of the degradation of the articulation of received speech voice in conventional circuits and permits pleasant communications at places where the background noise level is high. The circuit has a construction in which an input signal from a microphone is attenuated in correspondence to the background noise level to form a sidetone signal and a received speech signal from a speech channel is amplified in correspondence to the background noise level to form a new received speech signal.

Abstract: An automated method for modifying a speech signal communicated over a telephone network by applying a gain factor in a telephone set before any side tone is added thereto. The gain factor is a function of a received signal indicative of the background noise at the location of the telephone set. The gain applied may be a function of both the background noise level and the original speech signal. Either a linear or a non-linear (e.g., compressed) amplification of the original speech signal may be performed, where a compressed amplification results in the higher level portions of the speech signal being amplified by a smaller gain factor than lower level portions. The speech signal may be separated into a plurality of subbands, each resultant subband signal being individually modified in accordance with the present invention.

Abstract: A sidetone optimization circuit (230) provides the capability to adjust sidetone level received at a telephone handset to a desired level. A noise source (204) generates a noise signal that gets filtered by filter (206) in order to provide band limited noise to a telephone interface circuit (202). Telephone interface circuit (202) provides a sidetone signal (210), also known as a trans-hybrid loss signal. The trans-hybrid loss signal is filtered at filter (212) and transformed to a DC level at detector (214). The DC level is then compared to a predetermined threshold at comparator (224), and a variable impedance circuit (218) is adjusted until the DC level reaches the predetermined threshold in order to achieve a desired level of sidetone.

Abstract: An electronic hybrid useful in telephone line-powered Data Access Arrangements. The hybrid has a transmit amplifier with an output driving the base of a bipolar transistor. The emitter of the transistor is connected to a sense resistor which is, in turn, connected to one wire of a telephone line. The collector of the transistor connects to another wire of the telephone line. A summing node couples together signals on the telephone line and signals across the sense resistor such that the remaining signal on the summing node is substantially the signal received from the telephone line without the signal to be transmitted.

Abstract: A telephone line-powered Data Access Arrangement having amplifiers therein directly powered from the telephone line without voltage regulators or filters. The amplifiers, such as the amplifier driving an LED in an optical isolator, draw a constant power supply current independent of the signal being amplified. Thus, amplified signals are not coupled back onto the telephone line.

Abstract: An automated method for modifying a speech signal communicated over a telephone network by applying a gain factor in a telephone set before any side tone is added thereto. The gain factor is a function of a received signal indicative of the background noise at the location of the telephone set. The gain applied may be a function of both the background noise level and the original speech signal. Either a linear or a non-linear (e.g., compressed) amplification of the original speech signal may be performed, where a compressed amplification results in the higher level portions of the speech signal being amplified by a smaller gain factor than lower level portions. The speech signal may be separated into a plurality of subbands, each resultant subband signal being individually modified in accordance with the present invention.

Abstract: An automated method for modifying a speech signal in a telephone network by applying a gain factor which is a function of the level of background noise at a given destination, and transmitting the modified speech signal to the destination. The gain applied may be a function of both the background noise level and the original speech signal. Either a linear or a non-linear (e.g., compressed) amplification of the original speech signal may be performed, where a compressed amplification results in the higher level portions of the speech signal being amplified by a smaller gain factor than lower level portions. The speech signal may be separated into a plurality of subbands, each resultant subband signal being individually modified in accordance with the present invention. In this case, each subband speech signal is amplified by a gain factor based on a corresponding subband noise signal, generated by separating the background noise signal into a corresponding plurality of subbands.

Abstract: A telephone privacy device includes first and second connectors that are mateable with a telephone handset and a telephone base, respectively. The handset includes a microphone, and the device includes a variable amplifier that allows the user to amplify the microphone signal to a desired level prior to sending it on to the base. A signal strength display allows the user to monitor the amplitude of the amplified microphone signal. A side tone suppression circuit generates a side tone suppression signal in response to the amplified microphone signal. This side tone suppression signal is used to reduce excessive side tone from the telephone base. An adjustable gain amplifier amplifies the speaker signal supplied by the base for the handset speaker.

Abstract: An automated method for modifying a speech signal in a telephone network by applying a gain factor which is a function of the level of background noise at a given destination, and transmitting the modified speech signal to the destination. The gain applied may be a function of both the background noise level and the original speech signal. Either a linear or a non-linear (e.g., compressed) amplification of the original speech signal may be performed, where a compressed amplification results in the higher level portions of the speech signal being amplified by a smaller gain factor than lower level portions. The speech signal may be separated into a plurality of subbands, each resultant subband signal being individually modified in accordance with the present invention. In this case, each subband speech signal is amplified by a gain factor based on a corresponding subband noise signal, generated by separating the background noise signal into a corresponding plurality of subbands.

Abstract: To improve the anti-sidetone of electro-acoustic installations, particularly to improve the intercom characteristics of a telecommunication terminal, with compander control by an echo evaluation circuit (52) assigned to the compander (4), the latter has an echo compensator (40) located upstream, and/or a voice-controlled balance (51) is located downstream of the compander (4), which are controlled by the echo evaluation circuit (52) of the compander (4). To control the echo compensator (40) and/or the voice-controlled balance (51), the usual control circuitry required for their operation can be omitted. The result is a particularly effective, economical method with very good intercom characteristics.

Abstract: Selectively compressing the signal supplied to a loudspeaker based upon the volume setting for the sound level to be produced by the loudspeaker improves the intelligibility of human speech produced by a loudspeaker at high volume levels. Compression of the signal to the loudspeaker prevents the loudspeaker from overloading. The level of compression may be adjusted based upon the volume setting. The amplitude of an audio signal sent to a loudspeaker may be increased if below a preselected pivot point and increased if above the pivot point. The amount of increase or decrease in the amplitude may be proportional to the difference between the amplitude of the audio signal and the pivot point. This difference may also be raised to the power of a shape factor to determine the amount of increase or decrease in the amplitude of the audio signal.

Abstract: An attended messaging system having an audio message recording and playback device connected in series between a telephone base unit and a telephone receiver handset or headset. The recording and playback device includes a solid-state audio memory chip for storing a message and pairs of operational amplifiers connecting the memory chip of the recoding and playback device with the telephone base unit and with the speaker and microphone of the telephone receiver handset or headset in a differential circuit of audio signal channels. The recording and playback device is manually-actuable by means of buttons or switches to operate in a recording mode, in which a message spoken into the handset or headset microphone is stored in the audio memory chip, or in a playback mode, in which the previously recorded message is communicated through the telephone base unit and telephone line to a designated remote telephone station with which audio communication has been established by the telephone base unit.

Abstract: Speech signals appearing at the receiving and transmitting paths of a speakerphone which operates under voice switching, are received at a control circuit. The control circuit issues a control signal which assumes a first level in the event that a signal at the receiving path exceeds a signal at the transmitting path and which assumes a second level if the signal at the receiving path falls below the signal at the transmitting path. Communication loss is inserted in both the receiving and transmitting paths in response to the first level of the control signal, and is eliminated therefrom in response to the second level of the control signal.

Abstract: A remote simultaneous interpretation system particularly suitable for oral communication between a doctor and patient who do not speak a common language includes a user station equipped with a first and second headsets and microphones, a remote simultaneous interpretation system also equipped with a headset and a microphone and an audio link operatively interconnecting the stations. The simultaneous interpreter station includes controls for routing incoming and outgoing signals from and to separate portions of the user station, thereby allowing the remotely located interpreter to listen to either the doctor, the patient or both and to route voiced interpreter signals to the doctor, the patient or both. The use of simultaneous interpretation reduces the language barrier between a doctor and a patient, thereby improving rapport and overcoming social and/or cultural barriers to providing medical care.

Abstract: An automatic loss compensation for use in a two to four wire hybrid converter in a telephone system. Many hybrid converters provide a substantially fixed gain for receive signals that are provided to the customer loop telephone line and for transmit signals that are provided to the digital carrier transmission line. The compensation circuit includes a measurement circuit, controller, and receive and transmit amplifier. The measurement circuit determines the length of the customer loop telephone line and provides a length signal. The controller receives the length signal and categorizes the length as being within one of a plurality of segments. The controller appropriately varies the amplification and phase of the receive signal, which is then combined with an inverted transmit signal, in order to substantially reduce reflected signals from the transmission line and transmission line termination.

Abstract: A received signal vector x(n), a coefficient error covariance matrix P(n) from a memory part 12 and a forgetting factor .nu. from a memory part 13 are provided to a gain calculating part 14 to obtain a gain vector k(n). The thus obtained gain vector k(n) and an error e(n) between an echo and an echo replica are multiplied in a multiplying part 16. The multiplied output and a filter coefficient h(n) from a memory part 18 are added together to update the latter. The thus updated filter coefficient is used as the filter coefficient of an estimated echo path (an FIR filter, for example). The coefficient error covariance matrix P(n), the gain vector k(n), the received signal vector x(n) and the forgetting factor .nu.