Abstract: Methods and apparatuses for reducing effects of impulse noise in a DSL transmitter receiver device are described. According to certain embodiment, the method includes comparing a hard decision output of a decoder with a soft decision output for a convolution coded modulation symbol received at a digital subscriber line (DSL) receiver. The presence of impulse noise is detected based on a lack of agreement between the hard decision output and the soft decision output.

Abstract: A robust packet arrival time detector using a power estimate to validate a packet arrival time measurement. A packet arrival time measurement is considered valid when the value of the power estimate signal indicates that a packet is being received. A power estimator comprises a bandpass filter, a Hilbert transform, two squaring devices, an adder, and a lowpass filter.

Abstract: An extended bandwidth HomePNA system uses a transmit spectrum having a greater bandwidth than the bandwidth specified by a HomePNA 2.0 communication standard. The extended bandwidth system of the invention provides for additional copies of a spectrum of a 2 Mbaud training signal for better accuracy in decoding transmitted data by a receiver. In one embodiment, the extended bandwidth is a 12 MHz band centered at 10 MHz, from 4 MHz to 16 MHz. This extended bandwidth allows for three copies of a 4 Mbaud training sequence or six copies of a 2 Mbaud training sequence. The extended bandwidth of the invention is compatible with a HomePNA 2.0 system by providing a training sequence that enables a HomePNA 2.0 receiver in 2 Mbaud mode to train on the transmitted signal and determine that the transmitted signal is not intended for the HomePNA 2.0 receiver in 2 Mbaud mode.

Abstract: A detector for detecting packet arrival time for packets received by a Home Phoneline Network Alliance (HPNA) receiver. The detector correlates received preamble symbols with stored preamble symbols and generates an estimation of the packet arrival time when the entire transmitted preamble has arrived at the receiver. The estimation of the packet arrival time is defined on a symbol period boundary. The detector includes a complex correlator with a simplified structure based on characteristic of the received HPNA signal.

Abstract: A method and device for generating a constellation of Pulse Code Modulation (PCM) levels to represent digital data in a communication system. The constellation of Pulse Code Modulation levels are selected to have a total number of PCM levels and an upper power limit that optimizes noise performance. PCM levels are selected to increase the minimum spacing between PCM levels as great as possible while staying within the upper power limit for a constellation having a desired number of PCM levels. To minimize the power level of the selected constellation, constellation points are selected starting from PCM levels at linear values near zero to PCM levels at higher linear values. To improve noise performance, the number of occurrences of the minimum distance between PCM levels is also reduced without exceeding the upper power level. To reduce occurrences of the minimum distance between constellation points, PCM levels at the minimum distance from the previously selected PCM level may be skipped.

Abstract: An extended bandwidth HomePNA system uses a transmit spectrum having a greater bandwidth than the bandwidth specified by a HomePNA 2.0 communication standard. The extended bandwidth system of the invention provides for additional copies of a spectrum of a 2 Mbaud training signal for better accuracy in decoding transmitted data by a receiver. In one embodiment, the extended bandwidth is a 12 MHz band centered at 10 MHz, from 4 MHz to 16 MHz. This extended bandwidth allows for three copies of a 4 Mbaud training sequence or six copies of a 2 Mbaud training sequence. The extended bandwidth of the invention is compatible with a HomePNA 2.0 system by providing a training sequence that enables a HomePNA 2.0 receiver in 2 Mbaud mode to train on the transmitted signal and determine that the transmitted signal is not intended for the HomePNA 2.0 receiver in 2 Mbaud mode.

Abstract: Apparatus and methods of compensating for distortions introduced in communication signals received over a telephone network. An adaptive linear equalizer (LE) pre-filter is used for filtering incoming sampled communication signals received via the telephone network transmission medium. An adaptive decision feedback equalizer (DFE) is used in conjunction with the LE to provide error filtering as a hybrid equalizer system to determine which of a plurality of communication states of the sampled communication signals as an accurate estimate of the originally transmitted signals over the transmission medium, responsive to the filtered communication signals from the LE. An error signal is generated relative to the difference between the determined states of the decision mechanism of the DFE and the filtered communication signals from the LE.

Abstract: An improved DC compensation method for use in conjunction with telephony signalling. The method includes defining a frame that includes at least two n-bit codewords. An unsigned codeword is then identified within the frame by applying a rule to the codewords in the defined frame. Next, a sign bit is appended to the unsigned codeword, thereby producing a DC compensating codeword. The sign bit is selected based upon a weighting function of the linear values associated with previously transmitted codewords. The remaining unsigned codewords in the frame are assigned sign bits from user data.

Abstract: The present invention provides a method of generating a signal that may be used to determine the characteristic response of a communication channel that utilizes the public Digital Telephone Network (DTN). The channel includes the DTN, which may have Network Digital Attenuators (NDA) and/or Robbed Bit Signalling (RBS), and a Digital-to-Analog Converter (DAC), (also known as a codec), as well as the analog characteristics of the local loop, typically a twisted pair of copper wires. The present invention provides a method and apparatus to determine the optimal sampling instant of the received data stream. The present invention provides a probing signal that is well-suited for use in determining the channel's response to a known sequence of PCM codes used as data symbols. This is especially useful in so-called PCM modulation schemes that utilize the DTN, where knowledge of network and DAC distortion predicates the selection of available PCM codes used to represent data.

Abstract: A method of detecting the presence of signal impairments that introduce distortion in one or more time slots of a communication link by comparing a received training sequence of signal points with hypothetical received signal points, where the hypothetical points are predistorted versions of the training sequence, and are based on the known, predetermined training sequence of signal points. A detection matrix is formed where one dimension of the matrix corresponds to a time slot within a frame, preferably six time slots to a frame, and the other dimension corresponds to a particular impairment, or combination of impairments. The individual signal impairments are generally one or more of the following: robbed bit signaling where the robbed bit is replaced with a logic `0`; robbed bit signaling where the robbed bit is replaced with a logic `1`; robbed bit signaling where the robbed bit is replaced with a logic `1/2`; 3 dB digital attenuator; 6 dB digital attenuator.

Abstract: A method of encoding data into a digital sequence to be transmitted over the DTN so as to cause the DTN's codec to generate a multiple modulus M-ary signal in a manner that will facilitate efficient data transmission and recovery (decoding) by the distant end receiver, even in the presence of robbed-bit signaling (RBS). The preferred embodiments provide an apparatus and method of converting blocks of binary data to a corresponding block of M.sub.1 -ary, M.sub.2 -ary, . . . , M.sub.n symbols using Multiple Modulus Conversion (MMC) to maximize the data rate, while minimizing the required Signal-to-Noise Ratio (SNR) to achieve a desired error rate, in a system having a transmitter connected to the DTN via direct digital access and a receiver connected over a conventional analog subscriber line. A subset of codec codewords is used to represent the M-ary signals.

Abstract: A method and device for generating a constellation of Pulse Code Modulation (PCM) levels to represent digital data in a communication system. The constellation of Pulse Code Modulation levels are selected to have a total number of PCM levels and an upper power limit that optimizes noise performance. PCM levels are selected to increase the minimum spacing between PCM levels as great as possible while staying within the upper power limit for a constellation having a desired number of PCM levels. To minimize the power level of the selected constellation, constellation points are selected starting from PCM levels at linear values near zero to PCM levels at higher linear values. To improve noise performance, the number of occurrences of the minimum distance between PCM levels is also reduced without exceeding the upper power level. To reduce occurrences of the minimum distance between constellation points, PCM levels at the minimum distance from the previously selected PCM level may be skipped.

Abstract: An improved DC compensation method for use in conjunction with telephony signalling. The method includes defining a frame that includes at least two n-bit codewords. An unsigned codeword is then identified within the frame by applying a rule to the codewords in the defined frame. Next, a sign bit is appended to the unsigned codeword, thereby producing a DC compensating codeword. The sign bit is selected based upon a weighting function of the linear values associated with previously transmitted codewords. The remaining unsigned codewords in the frame are assigned sign bits from user data.

Abstract: A combined frame mapping technique that uses aspects of Minimum Modulus Conversion (MMC) and Shell Mapping (SM) to map data bits to a sequence of data symbols, or points. The apparatus includes a shell mapper to generate ring indices from K data bits and a modulus converter to select the signal points from within the ring based on B data bits. The apparatus and method permits the use of constellations having any integer number of points per ring. Variations in the constellations from time-slot to time-slot within a frame are also accommodated. The number of rings in each constellation remains constant, but the moduli vary. MMSM produces d.sub.min equal to the best of MMC and SM, and in some cases d.sub.min, may be better than that for either MMC or SM.

Abstract: The present invention provides a method of generating a signal that may be used to determine the characteristic response of a communication channel that utilizes the public Digital Telephone Network (DTN). The channel includes the DTN, which may have Network Digital Attenuators (NDA) and/or Robbed Bit Signalling (RBS), and a Digital-to-Analog Converter (DAC), (also known as a codec), as well as the analog characteristics of the local loop, typically a twisted pair of copper wires. The present invention provides a method and apparatus to determine the optimal sampling instant of the received data stream. The present invention provides a probing signal that is well-suited for use in determining the channel's response to a known sequence of PCM codes used as data symbols. This is especially useful in so-called PCM modulation schemes that utilize the DTN, where knowledge of network and DAC distortion predicates the selection of available PCM codes used to represent data.

Abstract: The present invention provides a method of generating a signal that may be used to determine the characteristic response of a communication channel that utilizes the public Digital Telephone Network (DTN). The channel includes the DTN, which may have Network Digital Attenuators (NDA) and/or Robbed Bit Signalling (RBS), and a Digital-to-Analog Converter (DAC), (also known as a codec), as well as the analog characteristics of the local loop, typically a twisted pair of copper wires. The present invention provides a method and apparatus to determine the optimal sampling instant of the received data stream. The present invention provides a probing signal that is well-suited for use in determining the channel's response to a known sequence of PCM codes used as data symbols. This is especially useful in so-called PCM modulation schemes that utilize the DTN, where knowledge of network and DAC distortion predicates the selection of available PCM codes used to represent data.

Abstract: A quadrature amplitude modulation (QAM) communication system is provided in which data can be communicated in any one of a plurality of QAM modes, such as 16-QAM, 32-QAM, and 64-QAM. A receiver detects the particular QAM mode transmitted on a trial and error basis, by attempting to decode the received data using different QAM modes until a synchronization condition is detected. The synchronization condition can require that a plurality of different synchronization tests be met. In a specific embodiment, a first synchronization test is met when a renormalization rate of a trellis decoder is below a threshold value. A second synchronization test is met when a first synchronization word is detected in the received data. A third and final synchronization test is met when a second synchronization word is detected in the received data. In order to reduce the cost of the receiver, most of the QAM mode dependent components are implemented using look-up tables stored in PROMs.

Abstract: A method and apparatus for communicating high definition television signals is provided. Video signals are divided into blocks of PCM data. The PCM data is processed using motion estimation and compensation to provide DPCM data. For each block, one of PCM and DPCM data is selected for transmission based on a predetermined criteria. The selected data is compressed using the discrete cosine transform to provide blocks of transform coefficients. The coefficients are quantized to improve their coding efficiency and variable length coded. The variable length coded coefficients are coded using a concatenated coding scheme with a Reed-Solomon outer code and trellis inner code to provide coded signals for transmission. The coded signals are transmitted using QAM.

Abstract: A method and apparatus are provided for adaptively equalizing data signals in a communications receiver. An unequalized data signal is demodulated. The demodulated data signal is filtered in an adaptive equalizer that initially updates adaptive filter coefficients using error signals derived from a first algorithm. A carrier lock signal is generated when a phase error of a filtered signal output from the adaptive equalizer reaches a threshold value. The adaptive filter coefficients are updated using error signals derived from a second algorithm instead of the first algorithm in response to the carrier lock signal. The first algorithm is a self-recovering equalization algorithm such as the Constant Modulus Algorithm. The second algorithm can be a decision directed algorithm. Carrier phase is recovered without the use of a phase rotator or phase de-rotator, by locating the adaptive equalizer inside of the carrier recovery loop.

Abstract: Convergence of a complex adaptive equalizer used in digital communications is substantially improved by updating all coefficients of the equalizer during each filter clock cycle. Complex signal data is passed through a plurality of successive delay stages to provide N sets of delayed complex signal data. The product of each set and a complex error signal is obtained. Each product is concurrently updated with previous product data for the set to provide N sets of updated complex coefficients for selective input to equalizer filters. In an illustrated embodiment, the updated coefficients are truncated and their gain is adjusted prior to input to the filters. The updated coefficients can be multiplexed to provide a clocked stream of coefficient sets for input to the equalizer filters. A VLSI implementation of the equalizer is also disclosed.