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A digital Class-D amplifier comprises a pulsewidth modulator (PWM) and an output stage. In this paper, we simplify the time-domain expression for the algorithmic PWM linear interpolation (LI) sampling process and analytically derive its double Fourier series expression. By means of our derivation, we show that the nonlinearities of the LI process are very low, especially given its modest computation complexity and low sampling frequency. In particular, the total-harmonic distortion (THD) ≈0.02% and foldback distortion is -98.4 dB (averaged from modulation indexes M=0.1 to 0.9) for the 4-kHz voiceband bandwidth @1-kHz input, 48-kHz sampling. We also describe a simple hardware for realizing the LI process. We propose a frequency doubler (with small overheads) for the pulse generator for the PWM, thereby reducing the counter clock rate by 2, leading to a substantial ∼47% power dissipation reduction for the Class-D amplifier. By means of computer simulations and on the basis of experimental measurements, we verify our double Fourier series derivation and show the attractive attributes of a Class-D amplifier embodying our simplified LI sampling expression and reduced clock rate pulse generator. We show that our Class-D amplifier design is micropower (∼60 μW @1.1 V and 48-kHz sampling rate, and THD ≈0.03%) and is suitable for practical power-critical portable audio devices, including digital hearing aids.