Designing a 2x interpolation filter for audio I want to perform 2x interpolation on an audio signal sampled at 44.1KHz by upsampling the signal by adding a zero after each original sample and then using a lowpass filter to interpolate the results.

Assuming an analogue elliptical filter (of order 10 or lower) as the basis for the design. What design specification would constitute a high quality audio interpolator that minimises aliasing?

For instance a passband and stopband at 0.45 * Nyquist and 0.55 * Nyquist respectively (Nyquist = 22.05KHz) with a stopband attenuation of -50dB and passband ripple of -0.1dB would be common for a low cost commercial DAC (I appreciate that digital to analogue conversion is a slightly different scenario). Would there be better values for doing high quality audio interpolation that minimises aliasing and doesn't truncate too many of the high frequencies before 22.05KHz.

For instance, is it a good idea to have the stop band above 22.05KHz for interpolation or does it not matter if the lowpass rolls off a little after 22.05KHz and what is acceptable?
@Keith: Here's the situation as I see it. Your original signal's spectrum is shown in panel (a) below. The solid lines are spectral energy and the dotted lines are spectral replications. After the zero-stuffing (upsampling by two), the stuffed sequence's spectrum is shown in panel (b). Notice the spectral image centered at 44.1 kHz that needs to be removed (attenuated). So after zero-stuffing you need to implement a digital filter whose passband cutoff freq is 0.45*22.05 kHz and whose stopband begins at 0.55*22.05 kHz. (I show that filter as the dashed line in panel (b).) But keep in mind, the filter’s Nyquist rate is 44.1 kHz (88.2/2 kHz) and not 22.05 kHz. The final spectrum you want is shown in panel (c).