  Digital pulse width modulation has been considered for high-fidelity and
high-efficiency audio amplifiers for several years. It has been shown that the
distortion can be reduced and the implementation of the system can be
simplified if the switching frequency is much higher than the Nyquist rate of
the modulating waveform. Hence, the input digital source is normally upsampled
to a higher frequency. It was also proved that converting uniform samples to
natural samples will decrease the harmonic distortion. Thus, in this paper, we
examine a new approach that combines upsampling, digital interpolation and
natural sampling conversion. This approach uses poly-phase implementation of
the digital interpolation filter and digital differentiators. We will show that
the structure consists of an FIR-type linear stage and a nonlinear stage. Some
spectral simulation results of a pulse width modulation system based on this
approach will also be presented. Finally, we will discuss the improvement of
the new approach over old algorithms.
