This paper introduces a computationally efficient, highly linear, natural sampling algorithm. The natural sampler is an essential block in a power efficient PWM digital audio amplifier. The algorithm is implemented on a DSP and comprises a prediction filter and an interpolator. The prediction filter uses feedback to get a very accurate estimate of the required sampling time for audio frequency signals. The interpolator extracts the PWM duty ratio from the input signal based upon the prediction filter's sampling estimate. The algorithm is scalable with performance requirements in terms of computation required as well as data and program memory.:

Authors: Midya, Pallab; Roeckner, Bill; Rakers, Pat; Wagh, Poojan
Affiliation: Motorola Labs, Schaumburg, IL
AES Convention: 109 (September 2000) Paper Number: 5194
Publication Date: September 1, 2000 Import into BibTeX
Subject: Analog Signal Processing
Permalink: https://www.aes.org/e-lib/browse.cfm?elib=9144

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