ffmpeg -y -i "input.wav" -af aresample=32000:cutoff=1 "output.wav"

 

it is not such type of filters as you might think, but of course a proper algorithm does some filtering.

modern DAWs are all very good, yet not identical.


an ideal SRC algorithm goes over the least common multiple.

for example the LCR of 96000 and 44100 is 14112000.

so if you want to export a 44100 file from a 96000 file, you first make 147 copies of every sample, and then press 320 of these samples into 1 sample of the target file.
the filtering is very easy and basically only consists of averaging these 320 samples, and the result is close to perfect. it is just very slow to process...

 

I'd like to add that what you said is true in principle for non-integer resampling, but that upsampling by some integer multiple (say 16x) and then, for efficiency's sake, interpolating up/down the remainder using (Mitchell, Catmull, B-Spline, ...) is more common in practice.