Performance of pure digital audio amplifiers using pulse width modulation (PWM) highly depends on the accuracy of the pulse-coded audio (PCM) signal to PWM modulation sequence conversion. This process implies the recovery of original analog signal values at irregular time instances bearing on a uniformly distributed PCM data only. The recovery, or “natural sampling”, requires interpolation processing giving a trade-off between accuracy of the result and computation speed. In this paper we propose a method for natural sampling providing tunable speed-performance constrains while giving the advantage of easy implementation in VLSI. Cubic polynomial interpolation and iterative solving algorithm, as well as experimental results, are presented in the paper.
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