An audio sampler (in "music" terms because I'm not experienced in dsp) is a software which loads an audio file and allows you to play it at different pitches by slowing it down and speeding it up.
The audio file should be playable at any new frequency, and the frequency at which I'm playing it can change at any time, even continuously for "pitch slides".
My understanding is that if I am pitching it up, I should use a lowpass filter first so I don't create frequencies above nyquist that will alias. After that I need to use an interpolation function.
What kind of filter I should use? IIR or FIR? What type? Please also keep in mind that for audio the phase changes and the ringing in the time domain can be undesired because they change the way the transients sound.