Resampling time series to regular array, then downsampling (Butterworth)
I am not sure why you concluded "A Butterworth Filter" is the way to go, and looks like you have come to the same question- what originally led you to that?

A Butterworth Filter implementation stems from copying analog filter techniques and from what I have learned with DSP is that you have the wonderful opportunity to NOT copy the analog. This is because analog design techniques have been derived from the mathematical limits of what standard R L and C components can give you, while in DSP you can more likely address the problem from an optimum mathematical perspective within the greater space of delays, adds and multiplies (to note and hopefully not confuse; delays, adds and multiplies are NOT strictly limited to digital implementations; for example SAW filters). This may be a good case in point if we look closer at why you would choose a Butterworth filter versus what optimized algorithms such as least squares (FIRLS in Matlab) can give you. The least squares algorithm will result in a linear phase FIR solution, with a stop band roll-off which is ideal for multi-rate resampling applications. fred harris has made a good case (I cannot find the specific reference) that for resampling applications, and FIR filter will always out-perform an IIR filter.

Further what is really great about FIRLS for multirate applications is that you can easily implement a multi-band filter, concentrating the rejection bands to be just at the alias locations, as in the example shown in the figure below showing the same filter design for either a decimation (DFIR) or interpolation (IFIR) design, in this case using a factor of 4:





To answer your question about upsampling: If the signal has been upsampled, this is typically done by inserting zeros (which creates all the aliases in frequency) followed by a low pass filter which will eliminate the aliases and in the process grow the zeros to their optimum interpolated value, so yes in this case, as long as the intepolation filter is present, you will not need an additional low pass filter. 

I show this in the following figure showing that if you do the interpolation first in a rational rate conversion (both interpolate and decimate) you can combine the IFIR and DFIR designs as one filter.



Also for resampling given its simplicity, consider using a CIC structure. If you are concerned about passband roll-off- the inverse Sinc compensator is quite simple (3 taps). Both are detailed in the following links:

Decimation concepts: Fast Integer 8 Hz 2nd Order LP for Microcontroller

Upsampling Spectrum: Upsampling and comb function

Inverse Sinc Filter: how to make CIC compensation filter

More about CIC: CIC Cascaded Integrator-Comb spectrum