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Featured researches published by Tim A. Williams.


international conference on acoustics, speech, and signal processing | 1987

A split control store VLSI for 32 kbps ADPCM transcoding

Luis A. Bonet; Tim A. Williams

This paper describes the architecture used in a 16 pin CMOS VLSI Digital Signal Processor which was designed by the authors to perform both ANSI and CCITT versions of the ADPCM standard. The part is designed to run from a 20 MHz clock source with an instruction cycle time of 100 ns. This design is a good example of the power of application specific DSP designs to reduce the cost of implementing stable algorithms.


global communications conference | 1989

ISDN U transceiver algorithm, development system, and performance

J. Girardeau; N. van Bavel; W. Kuenast; Shawn McCaslin; M. Pendleton; Tim A. Williams; S. Aly; B. Sayar; P. Hung; G. Koleyni; A. Deczky; P. Ferland; S. Goulet

Motorola and BNR are currently developing the MC145472, a single-chip implementation of a 2B1Q ANSI standard U-interface transceiver for ISDN (integrated services digital network) basic rate access (BRA). The transceiver provides full duplex transmission of 160 kb/s over two-wire nonloaded outside plant loops and is designed to meet the ANSI T1.601 1988 standards for use in both LT and NT1 applications. Its implementation involves state-of-the-art analog and digital signal processing (DSP) technology. The authors describe the functions and verifications of the transceiver, with emphasis on the DSP technology used for echo cancellation, equalization, and timing recovery; the design methodology and hardware testbed used to develop the transceiver algorithms; and performance results for the transceiver under both laboratory and field conditions.<<ETX>>


international conference on acoustics, speech, and signal processing | 1989

A systolic IIR decimator

Tim A. Williams

A discussion is presented of several methods of providing complex pole pairs in the transfer function of a decimator for oversampled converters. Utilization of pole cancellation with added compensating zeros yields a systolic structure that is easily implemented. Decomposition of the coefficient terms of the IIR decimator by means of a high radix modified Booths algorithm recording scheme is shown to be efficient. An extension to the Agarwal and Burrus transformation for poles close to the z=1 point on the z-plane is shown to allow a reduction in the precision of the coefficients.<<ETX>>


Archive | 1984

Logarithmic arithmetic logic unit

Tim A. Williams


Archive | 1984

Circuit for adding and/or subtracting numbers in logarithmic representation

Tim A. Williams


Archive | 1988

Multiple output oversampling A/D converter with each output containing data and noise

Nicholas R. van Bavel; Tim A. Williams


Archive | 1988

Oversampled A/D converter using filtered, cascaded noise shaping modulators

Nicholas R. van Bavel; Tim A. Williams


Archive | 1990

Data processor having split level control store

Luis A. Bonet; Tim A. Williams


Archive | 1988

Oversampled A/D converter having digital error correction

Nicholas R. van Bavel; Tim A. Williams


Archive | 1982

A time multiplexed n-ordered digital filter

Tim A. Williams

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