Abstract: The present invention overcomes all of the aforementioned problems by defining two upstream masks (U1, U2) and two downstream masks (D1, D2) and using a mask selectable system for the long reach digital subscriber line (LDSL), in which a unique modem feature is activated during handshake to automatically check for physical layer status in terms of spectral compatibility and, thus, automatically optimize the boosted mode with the use of the mask selectable system choose the best combination of upstream/downstream masks in any physical layer noise scenario.

Abstract: A method and system of the present inventions reduces both NEXT and FEXT interferences due to handshake tones into upstream and downstream neighboring services, both at the CPE and at the CO, when handshake is experienced. An embodiment of the present inventions is directed to significantly reduce the NEXT and/or FEXT interferences due to handshake tones. For example, NEXT and/or FEXT interferences due to G.994.1 handshake tones, identified as, bins 7 and 9 for the Upstream channel and bins 12, 14 and 64 for the Downstream channel may be reduced. An embodiment of the present inventions provides an algorithm that may be used for both NEXT and FEXT Handshake Interferences reduction at the CO and at the CPE. In addition, the algorithm may operate in the time domain and in the frequency domain. Frequency domain algorithm offers two options, both versions taking advantage of the high correlation time of the Handshake tone signals.

Abstract: A method and system of the present inventions reduces both NEXT and FEXT interferences due to handshake tones into upstream and downstream neighboring services, both at the CPE and at the CO, when handshake is experienced. An embodiment of the present inventions is directed to significantly reduce the NEXT and/or FEXT interferences due to handshake tones. For example, NEXT and/or FEXT interferences due to G.994.1 handshake tones, identified as, bins 7 and 9 for the Upstream channel and bins 12, 14 and 64 for the Downstream channel may be reduced. An embodiment of the present inventions provides an algorithm that may be used for both NEXT and FEXT Handshake Interferences reduction at the CO and at the CPE. In addition, the algorithm may operate in the time domain and in the frequency domain. Frequency domain algorithm offers two options, both versions taking advantage of the high correlation time of the Handshake tone signals.

Abstract: In accordance with one aspect of the invention, an apparatus is provided having a transmit path and a receive path for communicating data. The apparatus includes an analog to digital (A/D) converter that is disposed in communication with the receive path. The A/D converter operates at a first sampling rate, and it converts a received analog signal into a digital signal. The apparatus also includes a first echo canceller that is in communication with the receive path and that operates at the first sampling rate for estimating a first portion of an echo signal leaking from the transmit path to the receive path. The estimated echo signal is subtracted from the digital signal. The amplitude of the digital signal is increased by a digital gain. A decimator is disposed in communication with the receive path, whereby the decimator filters the digital signal that has a first sampling rate and emits a signal output at a reduced sampling rate.

Abstract: Approaches to generating clock signals are presented in which a signal is received from a line and a clock signal is generated as a function of the received signal. The received signal has portions that are correlated to a cyclostationary disturbance that is present on the line. Hence, by setting the rising edge and the falling edge of the clock as a function of the received signal may result in a clock signal that is substantially synchronous to the cyclostationary signal on the line.

Abstract: A method and system are provided for preventing data loss in a VoIP system. In particular, during a VoIP call, it is determined whether incoming ringing on a POTS line causes an unacceptable level of signal loss or errors. If so, for subsequent VoIP calls, the CO handling calls to the POTS line is instructed to either answer each call with a busy signal or automatically forward calls to the POTS line to the VoIP line or other selected telephone. Calling returns to normal upon ending of the VoIP call. In this manner, incoming ringing on the POTS line does not result in call dropping or lengthy retraining processes.

Abstract: Disclosed herein are various mechanisms for the adaptive cancellation of radio frequency interference (RFI) in Discrete Multitone (DMT)-based signal transmissions systems. various mechanisms for the adaptive cancellation of radio frequency interference (RFI) in Discrete Multitone (DMT) modulated signal transmissions systems. In at least one embodiment of the present invention, RFI introduced by one or more RFI disturbers (e.g., ham radios) into a DMT modulated signal may be detected by calculating an RFI canceller coefficient for some or all of the DMT bins of the signal, where the RFI canceller coefficient represents the correlation between the noise at the corrupted bin and the reference RFI noise. The RFI canceller coefficients for the corrupted bins may be calculated using an LMS adaptation process. Those bins where the magnitude of the corresponding RFI canceller coefficient is greater than a certain threshold may be marked as corrupted by RFI.

Abstract: This invention provides approaches for dual-modulation tuning in systems that exhibit self-disturbance. In one embodiment, a dual-bit-mapping scheme is used so that system parameters are set according to one bit-mapping scheme when self-disturbance is present and to another bit-mapping scheme when self-disturbance is not present in the system. Self-disturbance may include signals that arise from echoes or near-end cross-talk (NEXT).

Abstract: Systems and methods for improving data transmission are presented in which predefined masks are selected as a function of line conditions. The predefined masks may be time-frequency masks or power spectral density (PSD) masks. The predefined masks may be static masks, which do not adaptively change as a function of changing line conditions, or adaptive masks, which adaptively change as a function of changing line conditions. Sub-carriers are loaded with data as a function of the selected predefined mask.

Abstract: Systems and methods for increasing data capacity in communication systems is presented. In one embodiment, line characteristics are determined, and a downstream bandwidth is adaptively allocated in response to the determined line characteristics. In this regard, the upstream and downstream bandwidths may be seen as adaptively changing as a function of line characteristics.

Abstract: Systems and methods are presented for reducing noise induced by digital subscriber line (DSL) systems into services that are concurrently deployed on a communication line. In the disclosed technique, a power level of a discrete multi-tone (DMT) sub-carrier is adaptively calculated from a signal that is received from a communication line. The signal has information indicative of line conditions, which are further indicative of services deployed on the communication line.

Abstract: The present invention generally relates to echo cancellation over an asymmetric transmission and receiving spectra. An apparatus is provided having a transmit path and a receive path, with an adaptive echo canceler disposed therebetween, for communicating data within a first bandwidth and a second bandwidth respectively. A first decimator is disposed between the transmit path and the echo canceler circuit and filters an incoming signal having a first sampling rate and emits a signal output at a second, reduced sampling rate. A second decimator is disposed along said receive path and filters an incoming signal and emits a signal output at a reduced sampling rate. An adder is disposed to subtract the output of the adaptive echo canceler from output of the second decimator to generate a received signal that is substantially free of echo.

Abstract: An improvement to system timing and synchronization in a digital transceiver allows the synchronization of a transceiver located at a central office to a network or system clock without the use of any additional transmitted bits or additional external circuitry. The system allows a transceiver to achieve timing and synchronization lock to a system master clock, such as a T1 or E1 clock, by operating on the two level input clock with a complex rotator to develop an error signal which allows a phase locked loop circuit to drive a frequency synthesizing device, which in turn drives a system clock. The system clock provides an input to a complex generator, which in turn develops a complex rotation function. The complex rotation function allows the transceiver clock to synchronize to the network master clock.

Abstract: A system and method are disclosed for synchronizing a predetermined repeated data sequence contained in a first data signal at a first sample rate with the same predetermined repeated data sequence contained in a second data signal at a second sample rate in a data communications device. The system preferably includes a digital signal processor operating pursuant to logic stored on a memory. The logic includes synchronization logic which reduces the first sample rate to a predetermined correlation sample rate using a first decimator, resulting in a first reduced data signal. The synchronization logic also reduces the second sample rate to the same predetermined correlation sample rate using a second decimator, resulting in a second reduced data signal.

Abstract: An adaptive preceding system is implemented within a data communications system in order to precode data to track changes in a communications channel. The adaptive precoding system initializes a set of precoder values within a precoder filter of a transmitter during an initialization phase. The precoder filter is used to create a precoding signal that is combined with a transmission data signal in order to compensate for distortions on a communications channel. During a communications phase, a decision feedback equalizer generates equalizer coefficients that correspond to changes on the communications channel. The equalizer coefficients are periodically transmitted to a converter within the transmitter over a secondary channel. The converter determines a new set of precoder values that compensate for the changes in the communications channel and slowly updates the values in the precoder filter until they match the new set of precoder values.

Abstract: An improvement to a Reed Solomon (RS) coding scheme wherein the RS encoder and decoder is initiated based upon counting a number of timing, baud, or byte cycles from a known time stamp. The time stamp can be for example, a Tomlinson coefficient exchange frame whereby at the end of a Tomlinson coefficient exchange frame, a counter in both transmitter and receiver is actuated to begin counting a number of baud cycles. Alternatively, the counter may be initiated upon the receipt of a particular byte. Once the appropriate number of baud cycles or bytes have elapsed, the RS encoder and decoder will begin operation, thus ensuring that both RS encoding and decoding occur at the proper time, without the use of any additional framing bits.

Abstract: The invention provides a transceiver which is preferably used for high speed communications between a customer location and a central office over a subscriber line. The transceiver uses a pseudorandom noise generator both to provide a correlation for a complementary transceiver to use in achieving synchronization and as a way to transmit data in an extremely robust fashion. In addition, the transceiver uses frequency domain multiplexing so that downstream data and upstream data are transmitted by their respective transceivers in completely separate and distinct frequency bands. In order to achieve high data transfer rates, the transceivers can operate in an asymmetrical manner, whereby either (but, typically, the central office) transceiver can transmit data at a higher rate than the other transceiver.

Abstract: The problem of error propagation is resolved by using the communications channel to adapt a pre-equalizer of a transmitter, in response to changes in the communications channel. In particular, the pre-equalizer adapts to changes in the communications channel by processing an error signal that is communicated over a reverse channel by a corresponding receiver.

Abstract: A DSL communication device wherein the state of component coefficients are saved in a memory of a DSL device so that at a later time, the DSL device may initiate a warm start activation sequence wherein the last known channel parameters are used to train the equalizer, significantly reducing the amount of equalizer training required by eliminating the requirement that the aforementioned coefficients be transferred between DSL devices.

Abstract: The invention provides a transceiver which is preferably used for high speed communications between a customer location and a central office over a subscriber line. The transceiver uses a pseudorandom noise generator both to provide a correlation for a complementary transceiver to use in achieving synchronization and as a way to transmit data in an extremely robust fashion. In addition, the transceiver uses frequency domain multiplexing so that downstream data and upstream data are transmitted by their respective transceivers in completely separate and distinct frequency bands. In order to achieve high data transfer rates, the transceivers can operate in an asymmetrical manner, whereby either (but, typically, the central office) transceiver can transmit data at a higher rate than the other transceiver.