Zarlink Semiconductor Limited

Microsemi Semiconductor Limited

Mitel Corporation

Other

1021 Technologies Kk

Microsemi Semiconductor Ulc

Rim Semiconductor

Gary Q Jin

Gordon J Reesor

Thomas Qian

Dean Morgan

Jun Steed Huang

Yonghong Huang

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.

Method of adjusting a local clock in asynchronous packet networks

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&#x2266;M, adaptively filtered, and recombined to produce an output signal. The sub-bands are preferably oversampled such that N&#x3c;M. The use of oversampling and single-sided sub-banding reduces complexity and avoids aliasing problems.

Over-sampled single-sided subband echo cancellation

Disclosed is a method of reducing clicking sounds in an audio data stream. The samples of the audio data are delayed by a predetermined amount in a sliding window containing a predetermined number of samples. The presence of a clicking sound in the delayed data within said sliding window is detected and the corresponding audio data replaced by substitute data derived from the audio data stream.

Reduction of clicking sounds in audio data stream

An automatic gain control unit controls the gain applied to an input signal produced by a microphone subject to ambient noise. The automatic gain control circuit continually monitors the signal level of said input signal. A first gain control circuit decreases the gain applied to the input signal in increments of a first size when the input signal exceeds a first predetermined level. A second gain control circuit increases the gain applied to said input signal in increments of a smaller size when the input signal falls below a second predetermined level and in response to the presence of a speech present signal. The second predetermined level is less than said first predetermined level. In one embodiment, the first gain control circuit controls the analog gain of a codec and the second gain control circuit controls the digital gain of the codec.

Automatic gain control for mobile microphone

Timing information is transmitted over a network by sending time-stamped packets between a transmitter and receiver. The time-stamped packets are used to compute an estimate of the frequency deviation between the transmitter clock and the receiver clock. The local receiver clock is periodically updated to match the transmitter clock based on the estimate, which is preferably a least squares estimate. A multiple window approach is employed to increase update speed.

Method of transferring timing information over packet network

To reduce noise in an input signal that may contain speech, first an estimate of the noise level in the signal is obtained. The level of the input signal is then compared with the noise level estimate signal to determine whether speech is dominant. Less aggressive noise reduction is applied to the input signal when speech is dominant than when only noise is present.

Low complexity noise reduction method

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.

Phase locked loop with adaptive filter for DCO synchronization

In a method of effecting peak reduction in a DMT signal, a predetermined signature waveform is first produced. This is then subtracted from the DMT signal in the region of a signal peak whenever the DMT signal is above a predetermined maximum level.

DMT peak reduction without affecting transmission signal

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.

Efficient implementation of large size FFT

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 &#x3b3; to avoid numerical instability in the inverse matrix. The factor &#x3b3; is adaptively adjusted according to the characteristics of the signal.

Modified affine projection algorithm for non-stationary signal

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.

Flexible bit selection using turbo trellis-coded modulation

A method of determining cross channel interference in an Discrete Multitone (DMT) implementation of a Digital Subscriber Line (DSL) system. The cross channel interference is determined utilizing a residual impulse spectrum after implementation of a Time Equalization (TEQ) algorithm. In one application the cross channel interference value is used in a bit allocation algorithm to improve such that more bits are allocated to the channels with low interference and fewer bits are allocated to those channels having high interference. In this application the bit allocation algorithm is run twice, once before the interference measurement and once after.

DMT bit allocation with imperfect TEQ

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.

Soft-decision decoding of convolutionally encoded codeword

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.

Parallel turbo trellis-coded modulation

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.

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Kanata, Canada

Gary Q Jin