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An alternative method of digitizing music signals is presented. It relies on the techniques of spectral noise shaping and oversampling, which achieve theoretical and computed signal-to-quantization-noise ratios of 110-120 dB. This encoding scheme is based on high order delta-sigma modulation and is referred to as HLDSM A/D conversion. A hardware conversion of this system was developed measuring 80-dB SQNR and 0.007% total harmonic distortion. Potentially the HLDSM A/D converter offers noise and distortion performance superior to pulse-code modulation, a simpler hardware realization and reduced cost.
Author (s): Darling, Timothy F.;
Hawksford, Malcolm J.;
Affiliation:
University of California at Santa Barbara, Santa Barbara, CA ; University of Essex, Colchester, Essex, UK
(See document for exact affiliation information.)
Publication Date:
1990-12-06
DOI:
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Darling, Timothy F.; Hawksford, Malcolm J.; 1990; Oversampled Analog-to-Digital Conversion for Digital Audio Systems [PDF]; University of California at Santa Barbara, Santa Barbara, CA ; University of Essex, Colchester, Essex, UK; Paper ; Available from: https://aes.org/publications/elibrary-page/?id=6003
Darling, Timothy F.; Hawksford, Malcolm J.; Oversampled Analog-to-Digital Conversion for Digital Audio Systems [PDF]; University of California at Santa Barbara, Santa Barbara, CA ; University of Essex, Colchester, Essex, UK; Paper ; 1990 Available: https://aes.org/publications/elibrary-page/?id=6003
@article{Darling1990oversampled,
title={{Oversampled Analog-to-Digital Conversion for Digital Audio Systems}},
author={Darling, Timothy F. and Hawksford, Malcolm J.},
year={1990},
month={dec},
journal={Journal of the Audio Engineering Society},
volume={38},
number={12},
pages={924-943},
}
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