Abstract: The present invention provides, in one embodiment, a method of simplifying filtering with respect to a band plan. The method includes selecting a reflection point relative to the band plan and designating subchannels as unusable within a filterable separation of the reflection point that would otherwise introduce an asymmetry within the filterable separation. The present invention also provides a filtering simplifier for use with a DSL system having a band plan. The filtering simplifier includes a reflection point selector configured to select a reflection point relative to the band plan. The filtering simplifier also includes a subchannel designator coupled to the reflection point selector and configured to designate subchannels as unusable within a filterable separation of the reflection point that would otherwise introduce an asymmetry within the filterable separation.

Abstract: A number of basic equalization and demodulation structures have been shown to be appropriate for DMT systems depending on the channel, noise, and system parameters. These include single path, dual path, oversampled, and double rate structures. Using the fundamental computation units of two TEQs (FIR filters) and two FFTs, in conjunction with simple delays, downsampling and routing, single path, dual path, oversampled and double rate equalization structures can be realized from a common equalization structure.

Abstract: Method for computing distances to received data points. A preferred embodiment comprises determining a first point on a grid nearest to the received point, computing a second point closest to the received point inside a specified area, wherein the second point is a point in a first coset, computing a third, fourth, and fifth point, wherein each point is a member of a different coset and each point is the closest point in its coset to the received point, and computing a distance from the received point to each of the second, third, fourth, and fifth points.

Abstract: A method of providing an improved transfer function for a Discrete Multitone (DMT) type modulation transmitter with digital filtering after modulation followed by digital to analog converter and analog filtering is provided by sending a test signal to said transmitter and measuring the results of the test signal to determine the transfer function. The inverse of the transfer function that needs to be compensated is determined and the inverse of transfer function to be compensated is truncated to the region of interest (H2). The desired band split component of pre-compensation filter is designed (H3). The desired target frequency response of the pre-compensation filter is determined by H4=H2*H3 where H4 is the multiplication of H2 and H3 Given the target frequency response in H4 Hermetian symmetry is imposed on the frequency response. The inverse Fourier transform (IFFT) is taken to generate a time domain filter, h5. The characteristic of this filter is added at the digital filtering after modulation.

Abstract: A method is provided for accomplishing asymmetric digital subscriber loop classification and the design of passive hybrid networks for each of the classes. The resulting hybrids are suitable for implementation in a switchable hybrid architecture.

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

Abstract: The present invention provides automatic gain control (AGC) for a wide variety of conditions encountered in DSL scenarios. In particular, the proposed AGC approach addresses the problem of analog front end saturation in which a plurality of gain stages (i.e., programmable gain amplifiers) and filter stages are interleaved with inaccessible intermediate points. The gain settings of each of the PGAs is increased (614) until the received signal exceeds a given threshold (620) or until the maximum setting for that PGA has been reached (624). If the current PGA setting has been maximized, the next PGA is considered (628) and the process is repeated until no additional gain is needed or until the available PGAs are exhausted. This process is iterated (630) with different maximum PGA settings for each iteration to enable a more uniform gain distribution.

Abstract: The present invention provides an apparatus, system and method for synchronizing a local clock signal with a remote clock signal in a communication network. Phase information is used to calculate a number of “clock jitters” per unit of time needed to synchronize the locally generated clock with the remote clock. Introducing (removing) a given amount of delay at a particular point in the local clock signal results in a positive (negative) jitter in which its minimum value defines the jitter resolution. The jitters are introduced to the local clock signal from a plurality of tapped delay line elements (310) selected by a phase selector (350) in response to a timing correction signal issued by a phase error module (520).

Abstract: A method of a communicating across a channel includes receiving information having a known bandwidth and a known spectrum. The information is preferably in the form of a multicarrier modulated signal, e.g., a DMT signal. In one aspect, this information is received at a receiver having a reduced channel bandwidth. An aliasing spectrum can be calculated based on the known spectrum and the frequency difference between the known bandwidth and the reduced-channel bandwidth. The received information can then be modified based upon the aliasing function to compensate for alias distortion. For example, the received information can be modified by modifying the noise component or signal-to-noise ratio of the received information.

Abstract: The present invention provides a hardware assisted automatic gain control (AGC) for a communication network. A dedicated hardware portion of the AGC, which works in cooperation with software implemented functionality (400), is included to detect saturation conditions in the internal nodes of the analog front end (200) in which a plurality of gain stages (PGA1, PGA2, PGA3) and filter stages (H1, H2, H3) are interleaved with inaccessible intermediate points. The saturation detection logic includes a comparator (21, 22, 23) and flip-flop (27, 28, 29) for each gain stage (PGA1, PGA2, PGA3) and can be integrated directly in the analog front end 200. The dedicated hardware can further be included in a codec of a modem in a digital subscriber line (DSL) system.

Abstract: A modem operating in a narrow voltage range while maintaining a high signal to noise ratio during reception. The modem may contain a coder-decoder (CODEC) and a transformer. The CODEC receives data using more windings of a primary coil than the number of windings used for transmitting. As a result, the turns ratio is higher during transmission, leading to a correspondingly high amplification during transmission. The high amplification in the transmit direction enables the modem to operate in a narrow voltage range. As more windings of the primary coil are used for receiving, a signal of interest received from the telephone line is attenuated to a corresponding lesser degree, which leads to a high signal to noise ratio.