Abstract: In a computer-implemented method of adjusting a local clock at a receiver in a packet network, the local clock is generated by a phase locked loop locked to a master clock with the aid of time-stamped timing packets arriving over the network from the master clock with a packet delay distribution about a nominal delay. The timing packets are filtered to adjust for the packet delay distribution. A control input for the phase locked loop is derived from the timing packets. The amount of skew in the packet delay distribution about the nominal delay is determined, and the arrival times of timing packets are then selectively modified to correct for the amount of skew in the packet delay variation distribution prior to filtering the timing packets.

Abstract: A method of echo cancellation, particularly for use in high definition applications, splits an input signal and reference signal into M single-sided sub-band. The subbanded signals are downsampled at a downsampling rate N, where N?M, adaptively filtered, and recombined to produce an output signal. The sub-bands are preferably oversampled such that N<M. The use of oversampling and single-sided sub-banding reduces complexity and avoids aliasing problems.

Abstract: In a computer-implemented method of adjusting a local clock at a receiver in a packet network, the local clock is generated by a phase locked loop locked to a master clock with the aid of time-stamped timing packets arriving over the network from the master clock with a packet delay distribution about a nominal delay. The timing packets are filtered to adjust for the packet delay distribution. A control input for the phase locked loop is derived from the timing packets. The amount of skew in the packet delay distribution about the nominal delay is determined, and the arrival times of timing packets are then selectively modified to correct for the amount of skew in the packet delay variation distribution prior to filtering the timing packets.

Abstract: A method of echo cancellation, particularly for use in high definition applications, splits an input signal and reference signal into M single-sided sub-band. The subbanded signals are downsampled at a downsampling rate N, where N?M, adaptively filtered, and recombined to produce an output signal. The sub-bands are preferably oversampled such that N<M. The use of oversampling and single-sided sub-banding reduces complexity and avoids aliasing problems.

Abstract: In a closed loop acoustic system with forward and return paths, gain is controlled with different instantiations of a common gain control engine on each path. The gain control engine computes a first gain based on near-end signal levels, computes a second gain based on the output of the gain control engine, computes a third gain based on a howling condition, and computes a final gain as the product of the first, second and third gains.

Abstract: A digital phase locked loop has a digital controlled oscillator, a feedback loop coupled to the output of said digital controlled oscillator, a phase detector for comparing a feedback signal from said feedback loop with a reference signal to produce a phase error signal, and a low pass filter for filtering the phase error signal for controlling said digital controlled oscillator. A bandwidth calculation unit calculates the required filter bandwidth based on the phase error. The bandwidth calculation unit then controls the bandwidth of said low pass filter, which is thus adaptively adjusted in accordance with the phase error.

Abstract: A digital phase locked loop has a digital controlled oscillator, a feedback loop coupled to the output of said digital controlled oscillator, a phase detector for comparing a feedback signal from said feedback loop with a reference signal to produce a phase error signal, and a low pass filter for filtering the phase error signal for controlling said digital controlled oscillator. A bandwidth calculation unit calculates the required filter bandwidth based on the phase error. The bandwidth calculation unit then controls the bandwidth of said low pass filter, which is thus adaptively adjusted in accordance with the phase error.

Abstract: A system and method of implementing a Fast Fourier Transform (FFT) function in a high data rate communication network. The communication network, employing technology such as VDSL and DMT or FDM, frequently implements a FFT at a transmitter to transfer frequency domain modulated signals into time domain signals. An IFFT is implemented at the receiver to obtain the original signal. The present system divides the channel bandwidth into sub-bands and performs the FFT function with multiple FFTs in order to reduce chip size and computation time.

Abstract: An affine projection or like algorithm is used to process a non-stationary signal. The affine projection algorithm creates an inverse matrix and includes a factor ? to avoid numerical instability in the inverse matrix. The factor ? is adaptively adjusted according to the characteristics of the signal.

Abstract: An encoder structure for use in a DMT communications system, has a turbo encoder for encoding a portion of a data stream and generating a coded output. A data combiner has a first input receiving a remaining uncoded portion of the data stream and at least one further input receiving the coded output of the turbo encoder. A selector determines the portion of the data stream applied to the turbo encoder based on the data rate, latency requirement, coding gain performance and circuit complexity.

Abstract: A method and apparatus for decoding convolutional codes used in error-correcting circuitry for digital data communication. To increase the speed and precision of the decoding process, the branch and/or state metrics are normalized during the soft decision calculations, whereby the dynamic range of the decoder is better utilized. Another aspect of the invention relates to decreasing the time and memory required to calculate the log-likelihood ratio by sending some of the soft decision values directly to a calculator without first storing them in memory.

Abstract: An encoder for turbo coded trellis code modulation comprises an encoder data block for storing incoming data, and at least two recursive systematic convolutional encoders, said convolutional encoders being connected to receive data in parallel from said encoder data block. The decoder also employs a parallel implementation.

Abstract: A decision feedback equalizer is provided which is capable of canceling precursor Inter-Symbol Interference (ISI) even when knowledge of future data is unavailable. A novel DFE algorithm is utilized whereby the detection signal-to-noise ratio (SNR) is increased in two aspects: the precursor contribution is eliminated and the sampling location (relative to the channel response) is positioned at the maximum SNR location. As a result, the loop length for data transmission in a cable can be extended.

Abstract: In a method of transmitting data over a communications channel, at least some of the bits of an incoming bit stream are passed through a turbo encoder to generate turbo encoded output bits, and words corresponding to symbol points on a constellation in a trellis code modulation scheme are generated using at least the bits passed through the turbo encoder, possibly in conjunction with other bits that are not passed the through the turbo encoder. Typically, the turbo encoded bits are the least significant bits.

Abstract: A method of canceling echo signals in a telephone network operating in a double talk mode has been developed. A system for implementing the method is also presented. In the invention the conventional normalized least mean square (NLMS) algorithm for echo cancellers is modified such that echo path changes continue to be tracked even after a double talk condition has been detected. The modified algorithm, known herein as a double-talk normalized least mean square (DNLMS) algorithm adaptively adjusts the convergence rate based on the power difference between the echo signal or the residual signal and the far end signal.

Abstract: A method is disclosed for simplifying a Fast Fourier Transform operation on a signal that is real in the time domain, wherein advantage is taken of the symmetry in the frequency domain to reduce the number of butterfly operations required to derive the transform of the signal.

Abstract: An encoder structure for use in a DMT communications system, has a turbo encoder for encoding a portion of a data stream and generating a coded output. A data combiner has a first input receiving a remaining uncoded portion of the data stream and at least one further input receiving the coded output of the turbo encoder. A selector determines the portion of the data stream applied to the turbo encoder based on the data rate, latency requirement, coding gain performance and circuit complexity.

Abstract: A system and method of implementing a Fast Fourier Transform (FFT) function in a high data rate communication network. The communication network, employing technology such as VDSL and DMT or FDM, frequently implements a FFT at a transmitter to transfer frequency domain modulated signals into time domain signals. An IFFT is implemented at the receiver to obtain the original signal. The present system divides the channel bandwidth into sub-bands and performs the FFT function with multiple FFTs in order to reduce chip size and computation time.

Abstract: A decoder for use in a data communication system for decoding a stream of data which has been convolutionally and Reed-Solomon encoded is provided. The decoder has a trellis decoder for performing at least one iteration for decoding the stream of data. The Reed-Solomon decoder is provided for further decoding the encoded stream of data after the trellis decoder is stopped, the Reed-Solomon decoder including syndrome calculating means for calculating syndromes after each iteration of the trellis decoder. The decoder also includes control means for stopping the trellis decoder from performing another iteration when all of the syndromes calculated in the syndrome calculation means are zero.

Abstract: A method and apparatus for decoding convolutional codes used in error correcting circuitry for digital data communication. To increase the speed and precision of the decoding process, the branch and/or state metrics are normalized during the soft decision calculations, whereby the dynamic range of the decoder is better utilized. Another aspect of the invention relates to decreasing the time and memory required to calculate the log-likelihood ratio by sending some of the soft decision values directly to a calculator without first storing them in memory.