Agere Systems Inc.

Lucent Technologies Inc.

Agere Systems Guardian Corp.

Bahman Barazesh

Yhean-Sen Lai

Kannan Rajamani

Raja Banerjea

Zhenyu Wang

Jiangtao Xi

Tony S El-Kik

Steven C Szep

Geoffrey Lawrence Smith

Tomasz Janusz Wolak

Nitin Kumar Varma

Sherry Shuihong Zhu

San Hyok Yon

A unique real time tuning (RTT) process is employed for obtaining the desired optimum device parameter adjustments. The RTT parameter adjustment process is utilized with IP phone or other device chipsets as desired. In one embodiment, RTT provides a graphical user interface to a digital signal processor (DSP), or the like, on the device chipset allowing for observation, evaluation and control of the device parameters in real time. The real time exchange of the device parameter information between the device and an external workstation, e.g., a personal computer or the like, is provided by a User Datagram Protocol (UDP) that runs on a controller on the device, e.g., an ARM processor or the like. In this example, the unique combination of the RTT, UDP and DSP cooperate advantageously to implement, in accordance with the principles of the invention, the desired observability, and control to designers to tune the device, e.g., IP Phone, in real time to specified hardware, plastics, audio requirements required by existing standards or the like.

Real time tuner for providing graphical user interface

A circuit compensating for the difference in transmission rate of digital samples generated in transmit and receive paths between a user and a transceiver processing in the frequency domain, such as a digital multi-tone (DMT) transceiver. Compensation of the DMT transmission rate in the receive path in accordance with exemplary embodiments of the present employs zero-padding of the frequency domain coefficients generated by the DMT transceiver prior to applying an inverse transform, such as the inverse fast Fourier transform (IFFT). Zero-padding the frequency domain coefficients allows for the compensation of the transmission rate in the receive path by generating digital samples from the frequency domain coefficients with an inverse transform having a rate matched to the frequency domain transform and rate employed in the transmit path.

Transmission rate compensation for a digital multi-tone transceiver

A DMT signal conforming to a first DMT standard (e.g., the full-rate G.dmt standard based on 255 tones) is sampled at a sampling rate for the first DMT standard, filtered to attenuate a subset of the tones of the first DMT standard (e.g., all G.dmt tones above tone &num;127), and subsampled (e.g., 2:1) to provide a subsampled, filtered signal that can be further processed using components designed to operate under a second, different DMT standard (e.g., the half-rate G.lite standard based on 127 tones). As such, a conventional half-rate G.lite DMT transceiver can be modified (e.g., by changing the downstream sampling rate from 1.104 MHz to 2.208 MHz and adding an appropriate low-pass filter and decimator) for configuration in a full-rate G.dmt DMT system. The filtering and subsampling ensure that a downstream signal (even if it is a full-rate DMT initialization or synchronization signal containing tones above tone &num;127) can successfully be further processed using conventional half-rate DMT transceiver components, which are less complex and less expensive than those of full-rate DMT transceivers, thereby enabling the use of relatively inexpensive consumer personal equipment (CPE) in existing distributed full-rate DMT telecommunications systems.

Low-complexity DMT transceiver

Analog modems are enabled to better learn the slicing levels employed at the interface to a digital transmission network by reducing the effects of the various noise sources. Initially a training sequence is received to preliminarily adjust the analog modem's equalizer. Thereafter, a special training sequence, protected against intersymbol interference, is employed to collect samples of each slicing level, to ascertain the least mean squared value of each slicing level from the received samples and to obtain the channel's impulse response at each slicing level. Depending on the means squared error (MSE) between the input and the output of the slicer, the slicer tables continue to be updated, and the feed-forward and feed-backward equalizer filters are selectively adjusted in accordance with the channel impulse response ascertained at each of the slicing levels.

Apparatus and method for adapting a filter of an analog modem

Analog modems are enabled to better learn the slicing levels employed at the interface to a digital transmission network by reducing the effects of the various noise sources. Initially a training sequence is received to preliminarily adjust the analog modem's equalizer. Thereafter, a special training sequence, protected against intersymbol interference, is employed to collect samples of each slicing level, to ascertain the least mean squared value of each slicing level from the received samples and to obtain the channel's impulse response at each slicing level. To mitigate the effects of robbed bit signaling that may be employed in the digital transmission network, an array of slicers is provided to determine which bit position is being robbed and to base level learning on samples obtained from the non-robbed positions.

Equalizer training in the presence of network impairment

A so-called post equalization echo canceler is utilized in conjunction with transmitter and receiver data timing synchronization to enhance tracking of the echo path impulse response and convergence of the transversal filter in the post equalization echo canceler. This is realized by employing the equalization error in the receiver to adapt coefficients of the post equalization echo canceler transversal filter, in conjunction, with the transmitter and receiver data timing synchronization. The timing synchronization is realized by using sample rate conversion of the transmit sample rate to the receive sample rate and, in one example, variable phase interpolation of the converted timing signal. The receiver timing is recovered, and a phase error signal generated by the timing recovery unit is advantageously employed to adjust a variable phase interpolator in the receiver and a variable phase interpolator in a path supplying the transmitter signal to an input of the post equalization echo canceler. This insures that both the adaptive transversal filter of the post equalization echo canceler and a transversal filter in an equalizer in the receiver are operating on data having the same timing. In this example, the timing is that of the received data signal. In an embodiment of the invention, the post equalization echo canceler is utilized in conjunction with a so-called conventional, e.g., a primary, echo canceler. The conventional echo canceler is employed before the equalizer to cancel a major portion of any echo signal, while the post equalization echo canceler is employed after the equalizer to cancel residual echo signals caused primarily by drift in the hybrid network. To this end, the conventional echo canceler is &ldquo;trained&rdquo; during the initial half-duplex operation of the modem and, then, updating of its impulse response is inhibited, while the post equalization echo canceler is allowed to continue adapting.

Enhanced echo canceler

Analog modems are enabled to better learn the slicing levels employed at the interface to a digital transmission network by reducing the effects of the various noise sources. Initially a training sequence is received to preliminarily adjust the analog modem's equalizer. Thereafter, a special training sequence, protected against intersymbol interference, is employed to collect samples of each slicing level, to ascertain the least mean squared value of each slicing level from the received samples and to obtain the channel's impulse response at each slicing level. Thereafter, the analog modem's equalizer may be fine tuned in accordance with the channel impulse response ascertained at each slicing level.

Training of level learning modems

A multiple tone detector includes n tone detectors, each detecting one of n distinct tones, where n.gtoreq.2, and a background detector which generates a measure of accumulative background energy E.sub.avg in a frequency band or bands which do not include at least a subset of the n tones. The output of the background detector is applied to a smoothing filter, which generates the accumulative background energy measure E.sub.avg for a current frame as a weighted sum of the background detector output for the current frame and the background energy measure E.sub.avg from a previous frame. A parameter controlling response time of the smoothing filter is varied depending upon whether or not speech is determined to be present in the background portion of the input signal. A processor uses the energy measures from the n tone detectors and the background detector to compute n ratios, where a given ratio is the ratio of the energy measure of the ith tone to the accumulative background energy measure E.sub.avg. The processor determines if each of the n ratios are greater than a threshold, and if the maximum of the n ratios is less than a constant times the minimum of the n ratios, in order to generate a decision as to whether the n tones are present in the input signal.

Multiple tone detection using out-of-band background detector

A digital speech communication system having improved synchronization. The present digital speech communication system reduces the unit of degradation to a single speech sample, rather than a multi-sample frame, while maintaining the bit rate efficiency of the DSVD system and other systems where speech is encoded into large blocks and is subject to variable delay and mismatched clocks. The basic unit that is dropped or artificially inserted by the receiver, if the buffer overflows or empties, respectively, is reduced to a single speech sample. The speech frames produced by the demultiplexer are written into a frame buffer, in units of frames, at a rate determined by the clock signal, S2, that is extracted from the received signal by a timing recovery function in the modem. In accordance with the present invention, the frames are read out of the buffer into the decoder using the same extracted clock signal, S2. In this manner, once the buffer is partially full, the frame buffer will not overflow or empty. The speech decoder converts the coded speech into blocks of speech samples. The blocks of speech samples are then written to a variable frame buffer, in accordance with the extracted clock signal, S2. The variable frame buffer is allowed to partially fill, before the speech samples are read out to the digital-to-analog converter according to a clock signal, S3, at the 8 kHz sample rate, for presentation to the listener. When the variable frame buffer overflows, only a single speech sample needs to be discarded rather than an entire frame of multiple samples. Likewise, when the variable frame buffer empties, only a single extraneous speech sample need be inserted. The number of samples in the variable frame buffer will preferably be kept within predefined tolerances by a write process and a read process.

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Marlboro, NJ, United States of America

Bahman Barazesh